The driver bugs that caused decoded HEVC content to have an incorrect
memory layout have been fully fixed in the 410.xx driver release so
we can start exposing support.
This fixes the grammar of two HLS option descriptions and makes them less
ambiguous.
Signed-off-by: Werner Robitza <werner.robitza@gmail.com>
Signed-off-by: Lou Logan <lou@lrcd.com>
a thread count of 0 is treated the same as 1, use av_cpu_count() to get
the correct thread count when auto threads is requested.
this matches the fix in libvpxenc:
27df34bf1f avcodec/libvpxenc: fix setting amount of threads used for encoding
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: James Zern <jzern@google.com>
-1 will be map to error number "EPERM", and will be map to the error
message like "Error while decoding stream #0:0: Operation not permitted",
it's a strange error message when debug update_frame_pool fail,
now only return the error code from av_image_fill_pointers in case
of av_image_fill_pointers failure.
Signed-off-by: Jun Zhao <mypopydev@gmail.com>
In fmp4 & sub-range mode, the output's duration always smaller than expected,
because the size of the last #EXT-X-BYTERANGE is too small.
Signed-off-by: Charles Liu <liuchh83@gmail.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
The size of init.mp4 is zero in fmp4 mode,
when the input duraton smaller than the expected segment time.
fix ticket: 7166
Signed-off-by: Charles Liu <liuchh83@gmail.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
In fmp4 mode, the duration of the second m4s segment is
unusually smaller than the expected segment time.
Signed-off-by: Charles Liu <liuchh83@gmail.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Adds an option to specify the number of tile rows and columns, then uses
a uniform tiling if possible and otherwise a fixed tiling with equal-sized
tiles to fill the frame.
Also adds -tile-columns and -tile-rows options to make tilings with
power-of-two numbers of tiles, matching the behaviour of the libvpx/VP9
encoder.
This is needed because of 32bit float formats (which are difficult to
store in 16bits)
This also fixes undefined behavior found by fate
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
found_ref is not a single value in the bitstream. Fixes parsing files with
frame size changes.
Based on code from cbs_vp9.
Reviewed-by: Mark Thompson <sw@jkqxz.net>
Signed-off-by: James Almer <jamrial@gmail.com>
This adds common code to query driver support and set appropriate
address/size information for each slice. It only supports rectangular
slices for now, since that is the most common use-case.
This error isn't particularly helpful as checking for mixed IDR/non-IDR
NALUs would need to be done at a higher level to actually be accurate.
Removing the error allows an API user to send individual slice NALUs
(i.e. incomplete frames) so they can take advantage of slice
threading. The ticket which this error was added for (#4408) no
longer segfaults after removing this error (as the bug was likely
fixed more properly elsewhere).
Libx264 uses strtok which is not thread safe. Strtok is used in
x264_param_default_preset in param_apply_tune in x264/common/base.c.
Therefore the flag must be removed.
x264 fixed the issue, once the fix is pushed to stable, an #if can be added
to re-enable the flag based on X264_BUILD number.
Fixes ticket #7446.
Signed-off-by: Marton Balint <cus@passwd.hu>
This reverts commit f631c328e6.
The avcodec_parameters_to_context() call was freeing and reallocating
AVCodecContext->extradata, essentially taking ownership of it, which according
to the doxy is user owned. This is an API break and has produced crashes in
some library users like Firefox[1].
Revert until a better solution is found to internally propagate the filtered
extradata back into the decoder context, or a decision is made to change the
API.
[1] https://bugzilla.mozilla.org/show_bug.cgi?id=1486080
Signed-off-by: James Almer <jamrial@gmail.com>
Several SRT options are missing. Since pkg_config requires libsrt v1.3.0 and above, it should be able to support options added in libsrt v1.3.0 and below.
This commit adds 8 SRT options.
sndbuf, rcvbuf, lossmaxttl, minversion, streamid, smoother, messageapi and transtype
The keys of option are equivalent to stransmit.
https://github.com/Haivision/srt/blob/v1.3.0/apps/socketoptions.hpp#L196-L223
Signed-off-by: Marton Balint <cus@passwd.hu>
Allows arrangement of multiple windows such as:
ffmpeg -re -f lavfi -i mandelbrot -f sdl -window_x 1 -window_y 1 mandelbrot -vf waveform,format=yuv420p -f sdl -window_x 641 -window_y 1 waveform -vf vectorscope,format=yuv420p -f sdl -window_x 1 -window_y 481 vectorscop
Some changes by Marton Balint:
- allow negative position (partially or fully out-of-screen positions seem to
be sanitized automatically by SDL (or my WM?), so no special handling is
needed)
- only show window after the position is set
- do not use resizable and borderless flags at the same time, that caused
issues in ffplay
- add docs
Signed-off-by: Marton Balint <cus@passwd.hu>
Create SMPTE ST 12-1 timecodes based on H.264 SEI picture timing
info.
For framerates > 30 FPS, the field flag is used in conjunction with
pairs of frames which contain the same frame timestamp in S12M.
Ensure the field is properly set per the spec.
Currently qsv (m)jpeg encoding is broken.
Regression introducing by the commit(id: c1bcd3): fix async support,
which requires the minimum async_depth to be 1, instead previous zero.
But the default async_depth of qsv (m)jpeg encoding is still initialized
(mostly) as zero.
This patch also abviously improves qsv (m)jpeg encoding performance
due to the default async_depth is changed to 4.
Signed-off-by: Zhong Li <zhong.li@intel.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
64c50c0e97 declared support for decomposing
them but omitted to implement it; this adds an implementation.
Also do the same for end-of-stream NAL units, since they are equivalent.
case 1:
use the hexdump -C SMM0005.rcv get:
size skip (size - 4)
| |
V V
00000000 18 00 00 c5 05 00 00 00 4d f1 0a 11 00 e0 01 00
00000010 00 d0 02 00 00 0c 00 00 00 88 13 00 00 c0 65 52
^
|
size + 16
case 2:
same the command for SMM0015.rcv get:
size
|
V
00000000 19 00 00 c5 04 00 00 00 41 f3 80 01 40 02 00 00
00000010 d0 02 00 00 0c 00 00 00 00 00 00 10 00 00 00 00
^
|
size + 16
There are different the RCV file format for VC-1, vc1test
just handle the case 2 now, this fix will support the case 1.
(Both of test clips come from: SMPTE Recommended Practice -
VC-1 Decoder and Bitstream Conformance). And I think I got
a older VC-1 test clip in the case 1.
Reviewed-by: Carl Eugen Hoyos <ceffmpeg@gmail.com>
Reviewed-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Jun Zhao <jun.zhao@intel.com>
Signed-off-by: Yan, FengX <fengx.yan@intel.com>
Fixes bug with HTTP DELETE when HTTP Persistent is ON.
Right now, HTTP Persistent connections is supported only for POSTs and PUTs.
HTTP DELETE will still open a new connection every time.
Make the function static, or else Clang complains with:
error: no previous prototype for function 'decklink_get_attr_string' [-Werror,-Wmissing-prototypes]
Signed-off-by: Devin Heitmueller <dheitmueller@ltnglobal.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes: 10651/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ILBC_fuzzer-5202341540659200
Fixes: signed integer overflow: -1707705920 - 1703592888 cannot be represented in type 'int'
This tries to follow the webrtc code. For example using cliping and 64 bit as in WebRtcSpl_DotProductWithScale()
and not doing so in other places.
I could not find anything in rfc3951 and the reference code inside which would
explain what to do in these corner cases.
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
These limits are based on limiting done in WebRtcIlbcfix_CreateAugmentedVec()
Fixes: out of array accesses
Fixes: 10652/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ILBC_fuzzer-5638941487661056
Fixes: 10655/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ILBC_fuzzer-5699970020147200
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -2144033225 + -5208934 cannot be represented in type 'int'
Fixes: 10633/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RA_144_fuzzer-5679133791617024
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 7738135736989908991 - -7954308516317364223 cannot be represented in type 'long'
Fixes: find_stream_info_usan
Reported-by: Thomas Guilbert <tguilbert@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes trac issue #7215
Output for files created by xWMAEncode and various videogames is correct now.
1ch 32000hz files are still broken, would need fixes in WMA decoder.
Signed-off-by: bnnm <bananaman255@gmail.com>
Fixes crash noticed in the cbs_userdata patchset.
====ERROR: AddressSanitizer: heap-buffer-overflow on address 0x609000026c89 at pc 0x00010725d37b bp 0x7ffeea04e750 sp 0x7ffeea04e748
READ of size 4 at 0x609000026c89 thread T0
#0 0x10725d37a in ff_cbs_read_unsigned get_bits.h:274
#1 0x1072d2767 in ff_cbs_read_a53_user_data cbs_misc_syntax_template.c:119
#2 0x1078251a7 in h264_metadata_filter h264_metadata_bsf.c:595
#3 0x105c1321d in output_packet ffmpeg.c:853
0x609000026c89 is located 1 bytes to the right of 8-byte region [0x609000026c80,0x609000026c88)
allocated by thread T0 here:
#0 0x10aef08d7 in wrap_posix_memalign (libclang_rt.asan_osx_dynamic.dylib:x86_64h+0x578d7)
#1 0x10aca95e6 in av_malloc mem.c:87
#2 0x10ac545fe in av_buffer_allocz buffer.c:72
#3 0x107263b27 in cbs_h264_read_nal_unit cbs_h264_syntax_template.c:722
#4 0x10725b688 in cbs_read_fragment_content cbs.c:155
Signed-off-by: Aman Gupta <aman@tmm1.net>
The existing av_mediacodec_release_buffer allows the user to render
or discard the Surface-backed frame. This new method allows the user
to control exactly when the frame will be rendered to its SurfaceView.
Available since Android API 21.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Such streams are invalid according to
4.5.2.1 Top level payloads for the audio object types AAC main, AAC SSR, AAC LC and AAC LTP
4.5.2.1.1 Definitions
...cIn the raw_data_block(), several instances of the
same syntactic element may occur, but must have a different 4 bit
element_instance_tag, except for data_stream_element()'s and
fill_element()'s.
Fixes: Ticket7477
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Tool mediastreamvalidator reports error "Variant media_[N].m3u8 is
missing audio group" for audio streams in HLS master playlist. As audio
streams are already listed in audio group, skip them as variant media
streams in master playlist.
Libmfx requires 16 bytes aligned input/output for uploading.
Currently only output is 16 byte aligned and assigning same width/height to
input with smaller buffer size actually, thus definitely will cause segment fault.
Can reproduce with any 1080p nv12 rawvideo input:
ffmpeg -init_hw_device qsv=qsv:hw -hwaccel qsv -filter_hw_device qsv -f rawvideo -pix_fmt nv12 -s:v 1920x1080
-i 1080p_nv12.yuv -vf 'format=nv12,hwupload=extra_hw_frames=16,hwdownload,format=nv12' -an -y out_nv12.yuv
It can fix#7418
Signed-off-by: Zhong Li <zhong.li@intel.com>
RGB32(AV_PIX_FMT_BGRA on intel platforms) format may be used as overlay with alpha blending.
So add AV_PIX_FMT_BGRA format support.
One example of alpha blending overlay: ffmpeg -hwaccel qsv -c:v h264_qsv -i BA1_Sony_D.jsv
-filter_complex 'movie=lena-rgba.png,hwupload=extra_hw_frames=16[a];[0:v][a]overlay_qsv=x=10:y=10'
-c:v h264_qsv -y out.mp4
Rename RGB32 to be BGRA to make it clearer as Mark Thompson's suggestion.
V2: Add P010 format support else will introduce HEVC 10bit encoding regression.
Thanks for LinJie's discovery.
Signed-off-by: Zhong Li <zhong.li@intel.com>
Verified-by: Fu, Linjie <linjie.fu@intel.com>
This allows switching between absolute (LUFS) and relativ (LU) display
in the status line.
Signed-off-by: Daniel Molkentin <daniel@molkentin.de>
Signed-off-by: Conrad Zelck <c.zelck@imail.de>
This eases meeting the target level during live mixing.
Signed-off-by: Daniel Molkentin <daniel@molkentin.de>
Signed-off-by: Conrad Zelck <c.zelck@imail.de>
Allow to show short-term instead of momentary in gauge. Useful for monitoring
whilst live mixing.
Signed-off-by: Daniel Molkentin <daniel@molkentin.de>
Signed-off-by: Conrad Zelck <c.zelck@imail.de>
If we don't copy this value first, it is seen as 0 by h264_slice_header_init,
due to zero-allocation of the new context, triggering an old hack that
multiplied the denominator by 2 for files produced by old x264 versions, but
only if more than one thread was used.
Fixes#7475 and #7083.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This avoids surprising developers. Its bad to surprise developers with
such unexpected things.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
We check for the documented explanation of the "Ignore code" in extract_extradata_check() already
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This was the only case in the source that uses a hexadecimal shift value.
The change removed a special case in respect to greping
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes ticket #7441.
for block contrast calculate, the block is like this:
|<---------------- stride-----------------------|
+-----------------------------------------------> X
|
| w = 16
| (cx,cy)+------+
| | |
|h=blocksize| |
| | |
| +------+
V
Y
so we calc the block contrast use:
(cy + y) * stride + (cx + x)
Signed-off-by: Jun Zhao <mypopydev@gmail.com>
yae_set_tempo was overlooked when max tempo limit was raised to 100.
tested with:
./ffmpeg_g -i Delerium/SemanticSpaces/Gateway.mp3 \
-af asendcmd=f=asendcmd.cfg,atempo=1.0 -y /tmp/asendcmd-atempo.wav
where asendcmd.cfg was:
15.0-45.0 [enter] atempo tempo 2.0,
[leave] atempo tempo 0.5;
60.0-300.0 [enter] atempo tempo 4.0,
[leave] atempo tempo 1.0;
Simple parser to set keyframes, frame type, structure, width, height, and pixel
format, plus stream profile and level.
Reviewed-by: Mark Thompson <sw@jkqxz.net>
Signed-off-by: James Almer <jamrial@gmail.com>
That alone supports specifying the interface based on its address. Getting the
interface index from the local address seems quite a bit of work in a platform
independent way...
Obviously for IPv6 we still always use MCAST_JOIN_SOURCE_GROUP.
As a side effect this also fixes ticket #7459.
Signed-off-by: Marton Balint <cus@passwd.hu>
We already use localaddr for the multicast joins without source filters, so we
should use them for source filters as well. This patch only fixes the
IP_ADD_SOURCE_MEMBERSHIP and the IP_BLOCK_SOURCE case.
Unless we do this, the kernel automatically selects an interface based on the
source address, and that interface might be different from the one set in
localaddr. For blocked sources this even casues EINVAL because we joined the
multicast group on a different interface.
Signed-off-by: Marton Balint <cus@passwd.hu>
This allows getting data only from a specific source IP. This is useful not
only for unicast but for multicast as well because multicast source
subscriptions do not act as source filters for the incoming packets.
Signed-off-by: Marton Balint <cus@passwd.hu>
Variable 'ret' hasn't been initialized，thus introducing a random
hwupload failure regression due to qsv session uninitialized.
Signed-off-by: Zhong Li <zhong.li@intel.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This option is useful for maintaining input synchronization across N
different hardware devices deployed for 'N-way' redundancy.
The system time of different hardware devices should be synchronized
with protocols such as NTP or PTP, before using this option.
Signed-off-by: Marton Balint <cus@passwd.hu>
hevc parser mistakenly reports the following message if a dummy buffer
is padded for EOF
[hevc @ 0x559b63848610] missing picture in access unit
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Reviewed-by: "Li, Zhong" <zhong.li@intel.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Consider a component to be damaged if more than 50% of its subbands are damaged
Fixes: Timeout (part 1 of 2)
Fixes: 9774/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DIRAC_fuzzer-5748957085958144
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Divisions tend to be slower than shifts unless the compiler optimizes them out.
And some of these are in inner loops.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Because it will be used by avformat/segment.c or other module which
need to automatically create sub-directories operation.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
The value in AVCodecParameters->codec_tag may not be correct for IVF,
as it's the case when remuxing AV1 streams from mp4, so ignore it and
write the correct value based on codec ID instead.
Reviewed-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
It seems what this function does is a vertical prediction filter, thus
the new name should improve understanding.
rename the related table_b too
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Also bump the API version requirement to 10.9.5, because on olders versions
there were some reports of crashes using the undocumented, yet available
BMDDeckLinkDeviceHandle.
Signed-off-by: Marton Balint <cus@passwd.hu>
If vaEndPicture() failed in ff_vaapi_decode_issue(), free
the pic->slice_buffers.
Fixes the memory leak issue in ticket #7385
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Mark Thompson <sw@jkqxz.net>
Sets the level based on the stream properties if it is not explicitly
set by the user. Also add a tier option to set general_tier_flag, since
that affects the level choice.
Set profile compatibility/constraint flags properly (including the
constraint flags used for RExt profiles, as all streams we can currently
generate are RExt-compatible), and use that to add support for the "Main
Intra" and "Main 10 Intra" RExt subprofiles (for which we can re-use the
existing Main and Main10 VAAPI profiles).
A recent version of the standard changed the max and default to 15, from
16 in older versions. This updates the default to 15 to match, but the
max stays as 16 so that we don't reject older streams.
Give the entries in the VAAPI format map table an explicit type and add
functions to do the necessary lookups. Add another field to this table
indicating whether the chroma planes are swapped (as in YV12), and use
that rather than explicit comparisons where swapping is needed.
Clarify that the list is the naughty list, and therefore being on it is
not desirable. The i965 driver does not need to be on the list after
version 2.0 (when the standard parameter buffer rendering behaviour was
changed).
constraint_set1_flag should be set for constrained baseline and main
profiles, because the stream conforms to main profile.
constraint_set3_flag should be set for high profile when the stream
is intra-only.
constraint_set4_flag should always be set for main and high profiles
because interlaced encoding is not supported.
constraint_set5_flag should be set for main and high profiles when
B-frames are not used.
Also fix the setting of max_num_ref_frames - use the gop_size value
to check for intra-only rather than the constraint flag (which is not
necessarily set).
Add a larger warning more clearly explaining the consequences of missing
packed header support in the driver. Also only write the extradata if the
user actually requests it via the GLOBAL_HEADER flag.
Choose what types of reference frames will be used based on what types
are available, and make the intra-only mode explicit (GOP size one,
which must be used for MJPEG).
This was added in libva 2.1.0 (VAAPI 1.1.0). Use AVCodecContext.qmax,
matching the existing behaviour for qmin, and clean up the defaults so
that we only pass min/max when explicitly set.
Query which modes are supported and select between VBR and CBR based
on that - this removes all of the codec-specific rate control mode
selection code.
The codec sequence headers may contain fields which can overwrite the
fine parameters given in the specific settings (e.g. a crude bitrate
value vs. the max-rate / target-percentage / etc. values in
VAEncMiscParameterRateControl). Always reapply all global parameters
after a sequence header to avoid this causing problems.
Previously there was one fixed choice for each codec (e.g. H.265 -> Main
profile), and using anything else then required an explicit option from
the user. This changes to selecting the profile based on the input format
and the set of profiles actually supported by the driver (e.g. P010 input
will choose Main 10 profile for H.265 if the driver supports it).
The entrypoint and render target format are also chosen dynamically in the
same way, removing those explicit selections from the per-codec code.
Set the minimum version to 0.35.0 (libva 1.3.0) and remove redundant
configure tests. This also allows the proprietary libmfx fork of libva,
which always shows the version number 0.99.0 (independent of the actual
version).
Nothing prevents it to work except this check. AV1 is already supported
by Matroska muxer and aomenc produces WebM/AV1 files as well.
Signed-off-by: Kagami Hiiragi <kagami@genshiken.org>
Signed-off-by: James Almer <jamrial@gmail.com>
SIDX atom being inserted for every MOOF atom increases the muxing overhead.
This behaviour can be disabled for chunked CMAF format by enabling Global SIDX option of mov muxer.
Fixes: out of array read
Fixes: 10064/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-5766801384800256
Fixes: 10225/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-5071833448054784
Fixes: 10261/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-5115048024866816
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This optimization improved h264 decoding performance about 4%(from 74fps to 77fps, tested on loongson 3A3000).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This checks the value exactly for intra frames and checks it against a
minimum for inter frames as they can be variable.
Fixes: Timeout
Fixes: 10182/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ZMBV_fuzzer-6245951174344704
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
decomp_len is used in raw frames, so it should not be left at the value from
whatever was decoded previously (which may be any other frame)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Combined 1st and 2nd loop into one inline asm in function ff_vc1_inv_trans_8x8_mmi to
reduce memory operation, and made some small optimization in ff_vc1_inv_trans_4x8_mmi.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes high memory usage and prevents over allocation of the frames via
proper unref.
Can be checked as:
-hwaccel qsv -c:v h264_qsv -i ../h264-conformance/CANL2_Sony_E.jsv -c:v
h264_qsv -b:v 2000k -y qsv.mp4
For example bitdepth should be printed as 10 instead of 0A. Thanks to Hendrik Leppkes for pointing this out
Signed-off-by: James Almer <jamrial@gmail.com>
Failed case: svq3-watermark
When minimum loop count of following functions are greater than parameter h passed to them, svq3-watermark failed.
1. ff_put_pixels4_8_mmi
2. ff_avg_pixels4_8_mmi
3. ff_put_pixels4_l2_8_mmi
4. ff_avg_pixels4_l2_8_mmi
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes ticket #5654.
The linesize can be greater than the minimum required. This copies the
frame taking linesize into account.
Signed-off-by: Rick Kern <kernrj@gmail.com>
MSVC expands the preprocessor directives differently, making the
version check fail in the previous form.
Clang can warn about this with -Wexpansion-to-defined (not currently
enabled by default):
warning: macro expansion producing 'defined' has undefined behavior [-Wexpansion-to-defined]
Signed-off-by: Martin Storsjö <martin@martin.st>
The libaom doxy says that a value of 0 for the threads fields is
equivalent to a value of 1, whereas for avctx->thread_count it means
the maximum amount of threads possible for the host system.
Use av_cpu_count() to get the correct thread count when auto threads
is requested.
Reviewed-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
* commit '2edaafe5b93832715781851dfe2663da228a05ad':
libfdk-aacdec: Allow setting the new dynamic range control effect setting
Merged-by: James Almer <jamrial@gmail.com>
* commit 'ffb9b7a6bab6c6bfd3dd9a7c32e3724209824999':
libfdk-aac: Consistently use a proper version check macro for detecting features
Merged-by: James Almer <jamrial@gmail.com>
* commit '83678dbbae64ad8c501e0c732c1117e642c25dae':
libopenh264dec: Export the decoded profile and level in AVCodecContext
Merged-by: James Almer <jamrial@gmail.com>
* commit '8c76bfacf663ff71cee5264a74d0f9c86addd325':
tcp: Use ff_connect_parallel for RFC 8305 style connecting
Merged-by: James Almer <jamrial@gmail.com>
* commit '69caad8959982580504643d36aef22528e4aa6ce':
qsvdec: Release packet on decoding failure for mpeg2/vp8/vc1
Merged-by: James Almer <jamrial@gmail.com>
* commit 'e05e5920a4e1f1f15cc8a7c843159d519f6ec18e':
qsv: Error out if getting session handle failed in avfilter
Merged-by: James Almer <jamrial@gmail.com>
* commit '662558f985f50834eebe82d6b6854c66f33ab320':
decode: copy the output parameters from the last bsf in the chain back to the AVCodecContext
decode: flush the internal bsfs instead of constantly reinitalizing them
h264_redundant_pps_bsf: implement a AVBSFContext.flush() callback
vp9_superframe_bsf: implement a AVBSFContext.flush() callback
vp9_superframe_split_bsf: implement a AVBSFContext.flush() callback
h264_mp4toannexb_bsf: implement a AVBSFContext.flush() callback
bsf: add a flushing mechanism to AVBSFContext
This commit is a noop, see
b33f5299a5390f15645163e0846c66e9980c451e2954e5139394fe138de0f631c328e6
Merged-by: James Almer <jamrial@gmail.com>
* commit '6a9c00c09d2bc50c0ea64ba092b2f4afc46aa978':
tls_openssl: Fix checks for SSL_ERROR_WANT_WRITE in nonblocking operation
Merged-by: James Almer <jamrial@gmail.com>
* commit 'c194b9ad6dbe65f5abd68158c4811ed84e2a2b95':
network: Use ff_neterrno instead of AVERROR(errno) for poll errors
This commit is a noop, see 54b6bef6e1
Merged-by: James Almer <jamrial@gmail.com>
* commit '5d01bd181bb77e6740462095d7be4e0733a59420':
http: pass return code from http_open_cnx_internal() on its failure
This commit is a noop, see 70c9d40008
Merged-by: James Almer <jamrial@gmail.com>
* commit 'f89ec87afaf0d1abb6d450253b0b348fd554533b':
frame: Simplify the video allocation
Merged-by: James Almer <jamrial@gmail.com>
Padding-Remixed-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes bug id #7386
Muxer overhead calculations was intented for HLS playlist as Apple's mediastreamvalidator tests were failing.
But applying the same fix for DASH manifest proved counterproductive, as Bandwidth can be used for segment name templates.
Add missing dnxhr mxf container essence ULs to the mxf encoder.
This fixes dnxhr mxf files being quarantined by Avid Media Composer.
Signed-off-by: Jason Stevens <jay@wizardofthenet.com>
Reviewed-by: Baptiste Coudurier
refactor ff_dnxhd_get_hr_frame_size to avpriv_dnxhd_get_hr_frame_size,
to allow cross library usage in libavformat/mxfenc this change makes
this function no longer be always inlined.
Signed-off-by: Jason Stevens <jay@wizardofthenet.com>
When encoding to V210, make sure the AFD side data makes it through
in the resulting AVPacket. This is needed so the decklink output
module can put out AFD when in 10-bit mode.
Signed-off-by: Devin Heitmueller <dheitmueller@ltnglobal.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Create a new AVPacket side data type for Active Format Description,
which mirrors the side data type found in AVFrame. The primary
use case for this is ensuring AFD gets preserved in the V210
encoder, so that the decklink libavdevice can output AFD.
Signed-off-by: Devin Heitmueller <dheitmueller@ltnglobal.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Hook in libklvanc and use it for output of EIA-708 captions over
SDI. The bulk of this patch is just general support for ancillary
data for the Decklink SDI module - the real work for construction
of the EIA-708 CDP and VANC line construction is done by libklvanc.
Libklvanc can be found at: https://github.com/stoth68000/libklvanc
Updated to reflect feedback from Marton Balint <cus@passwd.hu>,
Carl Eugen Hoyos <ceffmpeg@gmail.com>, Aaron Levinson
<alevinsn_dev@levland.net>, and Moritz Barsnick <barsnick@gmx.net>
Signed-off-by: Devin Heitmueller <dheitmueller@ltnglobal.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
When encoding to V210, make sure the CC side data makes it through
in the resulting AVPacket. This is needed so the decklink output
module can put out captions when in 10-bit mode.
Signed-off-by: Devin Heitmueller <dheitmueller@ltnglobal.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
ISOBMFF does not allow AudioSampleEntryV1 in stsd version 0, so
assume the descriptor format is QTFF SoundDescriptionV1. ISOBMFF does
not define a version 2.
This fixes audio decoding for some MP4 files generated with Apple
tools. The additional fields present in SoundDescriptionV1/V2 need to
be read in order to correctly read additional boxes that contain
information required for decoding the stream.
Fixes#7376.
Also see: https://github.com/HandBrake/HandBrake/issues/1555
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Constraint "g" means compiler can store variable in memory or register.
When we use constraint "g" for a variable and this variable was operated by
instruction which only support register operands may lead "invalid operands" error.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It refers to the uncompressed quantization, therefore is not correct for AAC.
Also change mxf_set_pts to work based on current edit unit if
bits_per_coded_sample is not available.
Fixes error messages in the sample of ticket #7366.
Signed-off-by: Marton Balint <cus@passwd.hu>
Simplify the usage of intermediate variable addr and remove unused variable all64
in following functions:
1. ff_put_pixels_clamped_mmi
2. ff_put_signed_pixels_clamped_mmi
3. ff_add_pixels_clamped_mmi
This optimization speed up mpeg4 decode about 2% on loongson platform(tested with 3A3000).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The previous version checks checked explicitly for the version
where the version define was added to the installed headers,
making an "#ifdef AACDECODER_LIB_VL0" enough. Now that we have
a need for more diverse version checks than this, convert all checks
to such checks.
Signed-off-by: Martin Storsjö <martin@martin.st>
Simplify the usage of intermediate variable addr in following functions:
1. ff_put_pixels4_8_mmi
2. ff_put_pixels8_8_mmi
3. ff_put_pixels16_8_mmi
4. ff_avg_pixels16_8_mmi.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Removing unused VPP sessions by initializing only when used in order to help
reduce CPU utilization.
Thanks to Maxym for the guidance.
Signed-off-by: Joe Olivas <joseph.k.olivas@intel.com>
Signed-off-by: Maxym Dmytrychenko <maxim.d33@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Failed case: mss2-wmv
In following functions, pmullh was used to multiply two 16-bit data, this will cause data overflow.
1. ff_vc1_inv_trans_8x8_dc_mmi
2. ff_vc1_inv_trans_8x8_mmi
3. ff_vc1_inv_trans_8x4_mmi
4. ff_vc1_inv_trans_4x8_mmi
5. ff_vc1_inv_trans_4x4_mmi
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When flushing the encoder, we now need to provide non-null buffer
parameters for everything, even if they are unused.
The encoderDelay parameter has been replaced by two, nDelay and
nDelayCore.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'c011beda2611acfeb6f67d4fdf30d1eceed9e62f':
avconv: make sure packets put into the muxing FIFO are refcounted
This commit is a noop, see 33580a8625
Merged-by: James Almer <jamrial@gmail.com>
* commit 'b93026777aada7742583d8c5ab079e9f4dfe9a5d':
libfdk-aac: Use enum names instead of literal numbers for the output format
Merged-by: James Almer <jamrial@gmail.com>
* commit '52fd2afce8436c59c05765f3a6e95f9adb6f9f2f':
configure: fix inline asm checks
This commit is a noop, see ad94f1c8ab
Merged-by: James Almer <jamrial@gmail.com>
* commit 'f8060865f3e1a16c62e0d337ef0979b6ee4ba457':
qsvenc: use the compression_level to replace private option
Merged-by: James Almer <jamrial@gmail.com>
This requires us to pre-parse the skip data, as we want to
detect this before allocating all the arrays
Fixes: Timeout
Fixes: 9708/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMV2_fuzzer-5729709861109760
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'ad5bbc408637cffd4cc2ba990abef529cf5fa6a3':
configure: Rename require_header() --> require_headers()
This commit is a noop, see ce47f1589e
Merged-by: James Almer <jamrial@gmail.com>
* commit '4130e05ff496667565ff7c386a514bd46434eddf':
libavformat: add mbedTLS based TLS
This commit is a noop, see c24d247e2c
Merged-by: James Almer <jamrial@gmail.com>
* commit '39f3b6f3fc2b46b405b680cce3599f1b370e342d':
configure: Move add_fooflags() helper functions into canonical order
Merged-by: James Almer <jamrial@gmail.com>
* commit '5691c746cf62e69806aae1baf0a6e8252d519444':
configure: Group toolchain parameter mangling functions together
Merged-by: James Almer <jamrial@gmail.com>
* commit '25c2a27c9ec0150210d75ee5ac8ed1bfa14c1a56':
configure: Make require_cc() and require_cpp_condition() functions consistent
Merged-by: James Almer <jamrial@gmail.com>
* commit '78149d6657302b58d5e46e8bc0a521ed009f86f7':
amfenc: Retain a reference to D3D frames used as input during the encoding process
This commit is a noop, see 05f1a3face
Merged-by: James Almer <jamrial@gmail.com>
* commit 'abf806f7f1601c7e54de7f863bbb816af144a88c':
random_seed: use bcrypt instead of the old wincrypt API
This commit is a noop, see aedbf1640c
Merged-by: James Almer <jamrial@gmail.com>
* commit '347aa8f72356124ec6b95bf8ebd1faf72db03f8d':
x86: Don't declare a non-static function as inline
This commit is a noop
Merged-by: James Almer <jamrial@gmail.com>
Optimized memset with mmi in following functions:
1. ff_h264_add_pixels4_8_mmi.
2. ff_h264_idct_add_8_mmi.
3. ff_h264_idct8_add_8_mmi.
This optimization improved h264 decoding performance about 1.3%(tested on loongson 3A3000).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Reoptimize function ff_put_h264_chroma_mc8_mmi and ff_avg_h264_chroma_mc8_mmi.
Performance of h264 decoding improved about 5%(from 69fps to 73fps, tested on loongson 3A3000).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Performance of mpeg4 decoding improved about 23%(from 128fps to 158fps, tested on loongson 3A3000).
Reoptimized following functions with mmi.
1. ff_simple_idct_put_8_mmi
2. ff_simple_idct_add_8_mmi
3. ff_simple_idct_8_mmi
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Also make sure we set the URL context max packet size accordingly.
Based on a patch by Tudor Suciu <tudor.suciu@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Entries are always at least 8 bytes per the parsing code, so if we
see an impossible entry count avoid massive allocations. This is
similar to an existing check in mov_read_stsc().
Since ff_mov_read_stsd_entries() does eof checks, an alternative
approach could be to clamp the entry count to atom.size / 8.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
AV_CODEC_FLAG_GLOBAL_HEADER should be set before calling avcodec_open2() to have any effect.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
For cases with dual stack (IPv4 + IPv6) connectivity, but where one
stack potentially is less reliable, strive to trying to connect over
both protocols in parallel, using whichever address connected first.
In cases with a hostname resolving to multiple IPv4 and IPv6
addresses, the current connection mechanism would try all addresses
in the order returned by getaddrinfo (with all IPv6 addresses ordered
before the IPv4 addresses normally). If connection attempts to the
IPv6 addresses return quickly with an error, this was no problem, but
if they were unsuccessful leading up to timeouts, the connection process
would have to wait for timeouts on all IPv6 target addresses before
attempting any IPv4 address.
Similar to what RFC 8305 suggests, reorder the list of addresses to
try connecting to, interleaving address families. After starting one
connection attempt, start another one in parallel after a small delay
(200 ms as suggested by the RFC).
For cases with unreliable IPv6 but reliable IPv4, this should make
connection attempts work as reliably as with plain IPv4, with only an
extra 200 ms of connection delay.
Signed-off-by: Martin Storsjö <martin@martin.st>
- Allow to add deps in any order rather than "in linking order".
- Expand deps chains as required rather than just once.
- Validate that there are no cycles.
- Validate that [after expansion] deps are limited to other fflibs.
- Remove expectation for a specific output order of unique().
Previously when adding items to <fflib>_deps, developers were
required to add them in linking order. This can be awkward and
bug-prone, especially when a list is not empty, e.g. when adding
conditional deps.
It also implicitly expected unique() to keep the last instance of
recurring items such that these lists maintain their linking order
after removing duplicate items.
This patch mainly allows to add deps in any order by keeping just
one master list in linking order, and then reordering all the
<fflib>_deps lists to align with the master list order.
This master list is LIBRARY_LIST itself, where otherwise its order
doesn't matter.
The patch also removes a limit where these deps lists were expanded
only once. This could have resulted in incomplete expanded lists,
or forcing devs to add already-deducable deps to avoid this issue.
Note: it is possible to deduce the master list order automatically
from the deps lists, but in this case it's probably not worth the
added complexity, even if minor. Maintaining one list should be OK.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes the following warnings:
libavcodec/v4l2_m2m_enc.c:51:12: warning: missing braces around initializer
libavcodec/v4l2_m2m_enc.c:71:12: warning: missing braces around initializer
PTS is in microseconds, so correct field name is out_time_us.
Old field out_time_ms kept for now - will be removed after a suitable transition
period.
Fixes#7345
Just remove some dead variable assignments, unneeded variables and
change the FFMAX order to something more readable. Still identical.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Much simpler than regular decoding, does allow for 5.1 and 7.1
streams to be decoded without desync.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Unlike the range, the gradient start value does not have to be lower
than the end value.
Does allow more files to be correctly decoded without errors.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
x4 - x25 faster.
check_deps() recursively enables/disables components, and its loop is
iterated nearly 6000 times. It's particularly slow in bash - currently
consuming more than 50% of configure runtime, and about 20% with other
shells.
This commit applies few local optimizations, most effective first:
- Use $1 $2 ... instead of pushvar/popvar, and same at enable_deep*
- Abort early in one notable case - empty deps, to avoid costly no-op.
- Smaller changes which do add up:
- Handle ${cfg}_checking locally instead of via enable[d]/disable
- ${cfg}_checking: test done before inprogress - x2 faster in 50%+
- one eval instead of several at the empty-deps early abort path.
- The "actual work" part is unmodified - just its surroundings.
Biggest speedups (relative and absolute) are observed with bash.
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Tested-by: Helmut K. C. Tessarek <tessarek@evermeet.cx>
Tested-by: Dave Yeo <daveryeo@telus.net>
Tested-by: Reino Wijnsma <rwijnsma@xs4all.nl>
Signed-off-by: James Almer <jamrial@gmail.com>
x4 - x10 faster.
Inside print_enabled components, the filter_list case invokes sed
about 350 times to parse the same source file and extract different
info for each arg. This is never instant, and on systems where fork is
slow (notably MSYS2/Cygwin on windows) it takes many seconds.
Change it to use sed once on the source file and set env vars with the
parse results, then use these results inside the loop.
Additionally, the cases of indev_list and outdev_list are very
infrequent, but nevertheless they're faster, and arguably cleaner, with
shell parameter substitutions than with command substitutions.
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Tested-by: Helmut K. C. Tessarek <tessarek@evermeet.cx>
Tested-by: Dave Yeo <daveryeo@telus.net>
Tested-by: Reino Wijnsma <rwijnsma@xs4all.nl>
Signed-off-by: James Almer <jamrial@gmail.com>
x50 - x200 faster.
Currently configure spends 50-70% of its runtime inside a single
function: flatten_extralibs[_wrapper] - which does string processing.
During its run, nearly 20K command substitutions (subshells) are used,
including its callees unique() and resolve(), which is the reason
for its lengthy run.
This commit avoids all subshells during its execution, speeding it up
by about two orders of magnitude, and reducing the overall configure
runtime by 50-70% .
resolve() is rewritten to avoid subshells, and in unique() and
flatten_extralibs() we "inline" the filter[_out] functionality.
Note that logically, "unique" functionality has more than one possible
output (depending on which of the recurring items is kept). As it
turns out, other parts expect the last recurring item to be kept
(which was the original behavior of uniqie()). This patch preservs
its output order.
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Tested-by: Helmut K. C. Tessarek <tessarek@evermeet.cx>
Tested-by: Dave Yeo <daveryeo@telus.net>
Tested-by: Reino Wijnsma <rwijnsma@xs4all.nl>
Signed-off-by: James Almer <jamrial@gmail.com>
Encoder frame_number may be double-counted if some frames are cached and then flushed.
Take qsv encoder (some frames are cached firsty for asynchronism) as example,
./ffmpeg -loglevel verbose -hwaccel qsv -c:v h264_qsv -i in.mp4 -vframes 100 -c:v h264_qsv out.mp4
frame_number passed to encoder is double-counted and larger than the accurate value.
Libx264 encoding with B frames can also reproduce it.
Signed-off-by: Zhong Li <zhong.li@intel.com>
Fixes: signed integer overflow: -19818 + -2147483648 cannot be represented in type 'int'
Fixes: 9545/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SNOW_fuzzer-4928769537081344
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
ISMV lacks any sort of edit list support, as well as tfxd is
effectively the PTS of the fragment for most intents and purposes.
Thus, if b-frames are requested without negative CTS offsets you
end up with N frames' worth of delay (tfxd PTS plus the CTS offset
of the first sample). Negative CTS offsets enable the first sample
to have CTS=DTS, and thus a/v desync due to b-frame reorder delay
is avoided.
Since libopus 1.2, packets of sizes 80ms, 100ms and 120ms are allowed.
Fixes assertion failures when trying to mux such streams.
Signed-off-by: James Almer <jamrial@gmail.com>
Packets of sizes 80ms, 100ms and 120ms are allowed since libopus 1.2
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This reverts commit 7e0df5910e.
"complete frames" containers, even if they don't need to assemble
packets, still depended on this code for proper packet duration and
timestamp generation.
remove redundant av_init_packet after av_packet_unref.
av_packet_unref have call av_init_packet and reset the packet size.
Signed-off-by: Jun Zhao <mypopydev@gmail.com>
H264/265 have been fixed such an issue with commit
559370f2c4.
Similar fixing is needed for other codecs.
Signed-off-by: Zhong Li <zhong.li@intel.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Solve some issues found by an automated code scansion.
Suppress the complain "variables 'handle' is used but maybe
uninitialized".
Signed-off-by: Zhong Li <zhong.li@intel.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
If there is a saio/saiz in clear content, we shouldn't create the
encryption index if we don't already have one. Otherwise it will
confuse the cenc_filter.
The changed method is also used for senc atoms, but they should not
appear in clear content.
Found by Chromium's ClusterFuzz: https://crbug.com/873432
Signed-off-by: Jacob Trimble <modmaker@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
fix the waring: libavcodec/libkvazaar.c:210:27: warning: passing argument 3 of ‘av_image_copy’ from incompatible pointer type [-Wincompatible-pointer-types]
frame->data, frame->linesize,
^~~~~
In file included from libavcodec/libkvazaar.c:31:0:
./libavutil/imgutils.h:119:6: note: expected ‘const uint8_t ** {aka const unsigned char **}’ but argument is of type ‘uint8_t * const* {aka unsigned char * const*}’
void av_image_copy(uint8_t *dst_data[4], int dst_linesizes[4],
Signed-off-by: Jun Zhao <mypopydev@gmail.com>
fix the build warning for "ISO C90 forbids mixed declarations and code"
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Jun Zhao <mypopydev@gmail.com>
Currently float are converted to 16b uint in input part
using src depth (32 bits) in hScale16To19 and hScale16to15,
make an invalid shift for the data
So shift the value when using float input
like 16 bpc uint.
Add fix a memory leak issue as James's comments.
V2: use a local pict_type since coded_frame is deprecated.
Signed-off-by: Zhong Li <zhong.li@intel.com>
The specification states "NSV files may contain a single file header. "
Fixes: out of array access
Fixes: nsv-asan-002f473f726a0dcbd3bd53e422c4fc40b3cf3421
Found-by: Paul Ch <paulcher@icloud.com>
Tested-by: Paul Ch <paulcher@icloud.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is a temporary workaround for transcoding scenarious using libaom-av1
encoder, which currently can't propagate extradata during initialization.
Signed-off-by: James Almer <jamrial@gmail.com>
This fixes the creation of the hls manifest in hlsenc.c by writing the
entire manifest at the end for VOD playlists. Live & Event Playlists are unaffected.
This also fixes the behavior with HLS_TEMP_FILE to work correctly when
-hlsflags temp_file is specified, instead of always relying on use_rename, which caused these problems.
Files that would previously take over a week to fragment now take
1 minute on the same hardware. This was a 153 hour audio file (2.2GB of audio).
Signed-off-by: Ronak Patel <ronak2121@yahoo.com>
Fixes: long running loop
Fixes: ivr-timeout-42468cb797f52f025fb329394702f5d4d64322d6
Found-by: Paul Ch <paulcher@icloud.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This will get ISOBMFF and Matroska up to date with the revised AV1 Codec
Configuration Box spec.
For now keep propagating raw OBUs as extradata until all libavcodec modules
are adapted to handle AV1CodecConfigurationRecord formatted extradata.
Tested-by: Thomas Daede <bztdlinux@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Also only initialize it in ff_av1_packet_split() and not ff_av1_extract_obu(),
same as h2645_parse, so GetBitContext specific failures may not affect the
latter.
Signed-off-by: James Almer <jamrial@gmail.com>
Certain AVCodecParameters, like the contents of the extradata, may be changed
by the init() function of any of the bitstream filters in the chain.
Signed-off-by: James Almer <jamrial@gmail.com>
Initialize the bsfs once when opening the codec and uninitialize them once when
closing it, instead of at every codec flush/seek.
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: runtime error: left shift of 1 by 31 places cannot be represented in type 'int'
Fixes: 9480/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-6647324284551168 -rss_limit_mb=2000
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -540538872 + -2012739576 cannot be represented in type 'int'
Fixes: 9255/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-5758630052757504
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Previously, AVERROR(EIO) was returned on failure of
http_open_cnx_internal(). Now the value is passed to upper level, thus
it is possible to distinguish ECONNREFUSED, ETIMEDOUT, ENETUNREACH etc.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes vorbis mp4 audio files, with edit list specified. Since
st->skip_samples is not set in case of vorbis , ffmpeg computes the
start_time as negative.
Signed-off-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Neutrals are supposed to be anything not black (0,0,0) and not white
(N,N,N).
Previous neutral filtering code was too strict by excluding colors with
any of its RGB component maxed instead of just the white color.
Reported-by: Royi Avital <royiavital@yahoo.com>
Existing link is broken.
This patch updates the existing url with a working one.
Signed-off-by: Mina <minasamy_@hotmail.com>
Signed-off-by: Gyan Doshi <ffmpeg@gyani.pro>
Currently extra_hw_frames can't be applied to qsv since it
doesn't call function avcodec_get_hw_frames_parameters().
Give an option to fix ticket #7261 though it is not a perfect soultion
(allocate the minimum pool size internally and automatically).
Signed-off-by: Zhong Li <zhong.li@intel.com>
we need to make sure that memory allocation for Y/UV planes is continuous and re-used from a
pool
Signed-off-by: Maxym Dmytrychenko <maxim.d33@gmail.com>
When there is no metadata attached to a frame, take into account both
the PQ and HLG transfers, and change the HLG default value to 10:
the value of 12 is the maximum range in scene referred light, but
the reference OOTF maps this from 0 to 1000 cd/m² on the ideal HLG
monitor.
This matches what vf_tonemap_opencl does.
Alternatively the parser could be re implemented / redesigned so as to better
and more efficiently find frame boundaries
Fixes: Timeout
Fixes: 9210/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PBM_fuzzer-4770771833454592
Fixes: 9214/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PPM_fuzzer-5741633353023488
Fixes: 9219/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PGM_fuzzer-6249230237696000
Fixes: 9550/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PAM_fuzzer-5312669836902400
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: simple_idct_template.c:184:30: runtime error: signed integer overflow: -1065517056 - 1392182838 cannot be represented in type 'int'
Fixes: simple_idct_template.c:269:21: runtime error: signed integer overflow: 16384 * 259254 cannot be represented in type 'int'
Fixes: simple_idct_template.c:164:17: runtime error: signed integer overflow: 21407 * 210162 cannot be represented in type 'int'
Fixes: simple_idct_template.c:167:17: runtime error: signed integer overflow: 21407 * 210162 cannot be represented in type 'int'
Fixes: simple_idct_template.c:169:19: runtime error: signed integer overflow: 22725 * 259190 cannot be represented in type 'int'
Fixes: simple_idct_template.c:171:19: runtime error: signed integer overflow: 19265 * 259190 cannot be represented in type 'int'
Fixes: simple_idct_template.c:173:19: runtime error: signed integer overflow: 12873 * 259190 cannot be represented in type 'int'
Fixes: simple_idct_template.c:183:28: runtime error: signed integer overflow: 1860878336 + 585177665 cannot be represented in type 'int'
Fixes: simple_idct_template.c:159:17: runtime error: signed integer overflow: 16384 * 189520 cannot be represented in type 'int'
Fixes: simple_idct_template.c:170:22: runtime error: signed integer overflow: 19265 * 130147 cannot be represented in type 'int'
Fixes: simple_idct_template.c:174:23: runtime error: signed integer overflow: -22725 * 130147 cannot be represented in type 'int'
Fixes: simple_idct_template.c:183:20: runtime error: signed integer overflow: 16384 * -175206 cannot be represented in type 'int'
Fixes: simple_idct_template.c:184:22: runtime error: signed integer overflow: -16384 * -175206 cannot be represented in type 'int'
Fixes: simple_idct_template.c:185:22: runtime error: signed integer overflow: -16384 * -175206 cannot be represented in type 'int'
Fixes: simple_idct_template.c:186:20: runtime error: signed integer overflow: 16384 * -175206 cannot be represented in type 'int'
Fixes: simple_idct_template.c:195:26: runtime error: signed integer overflow: 19265 * 150747 cannot be represented in type 'int'
Fixes: simple_idct_template.c:198:27: runtime error: signed integer overflow: -22725 * 150747 cannot be represented in type 'int'
Fixes: simple_idct_template.c:184:37: runtime error: signed integer overflow: 21407 * -171941 cannot be represented in type 'int'
Fixes: simple_idct_template.c:185:37: runtime error: signed integer overflow: 21407 * -171941 cannot be represented in type 'int'
Fixes: simple_idct_template.c:192:27: runtime error: signed integer overflow: -12873 * 206341 cannot be represented in type 'int'
Fixes: simple_idct_template.c:185:30: runtime error: signed integer overflow: 1196441600 + 1703756981 cannot be represented in type 'int'
Fixes: simple_idct_template.c:176:23: runtime error: signed integer overflow: -12873 * 168461 cannot be represented in type 'int'
Fixes: simple_idct_template.c:191:27: runtime error: signed integer overflow: -22725 * -140062 cannot be represented in type 'int'
Fixes: simple_idct_template.c:197:26: runtime error: signed integer overflow: 19265 * -140062 cannot be represented in type 'int'
Fixes: simple_idct_template.c:183:34: runtime error: signed integer overflow: 8867 * -243046 cannot be represented in type 'int'
Fixes: simple_idct_template.c:186:34: runtime error: signed integer overflow: 8867 * -243046 cannot be represented in type 'int'
Fixes: simple_idct_template.c:186:28: runtime error: signed integer overflow: -816234496 - 2139878414 cannot be represented in type 'int'
Fixes: simple_idct_template.c:188:26: runtime error: signed integer overflow: 12873 * -239872 cannot be represented in type 'int'
Fixes: simple_idct_template.c:165:16: runtime error: signed integer overflow: 8867 * -260084 cannot be represented in type 'int'
Fixes: simple_idct_template.c:166:16: runtime error: signed integer overflow: 8867 * -260084 cannot be represented in type 'int'
Fixes: 9135/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-6324422955761664
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -1813244069 + -1407981383 cannot be represented in type 'int'
Fixes: 8823/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-5643295618236416
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This code came originally from gstreamer, where it was added in [1]
as a work-around for the Tegra 3. (The alignment was changed in [2]
as a response to [3], from 32-bit to 16-bit).
gstreamer only used this workaround in the case where the decoder
didn't return a slice-height property, but when the code was copied
into avcodec the conditional got lost. This commit restores the guard
and prefers the slice-height from the decoder when it is available.
This fixes segfaults decoding 1920x1080 h264 and mpeg2 videos on the
NVidia SHIELD after upgrading to Android Oreo.
[1] a870e6a5c3
[2] 21ff3ae0b0
[3] https://bugzilla.gnome.org/show_bug.cgi?id=748867
Signed-off-by: Aman Gupta <aman@tmm1.net>
Fixes some SVQ3 encoded files which fail to decode correctly after 6d6faa2a2d.
These files exhibit lots of artifacts and logs show "Media key encryption is not implemented".
However they decode without artifacts before 6d6faa2a2d.
The attatched patch allows these files to successfully decode, but also reject media key files.
Tested on the files in #6094 and http://samples.mplayerhq.hu/V-codecs/SVQ3/Vertical400kbit.sorenson3.mov
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Make sure to not write forbidden OBUs to CodecPrivate, and do the same with
unnecessary OBUs for packets.
Signed-off-by: James Almer <jamrial@gmail.com>
Some containers, like Matroska, may propagate key frames with no Sequence
Header OBU since it's provided in extradata instead.
With this change, the Sequence Header will be appended to the packet data
before calling aom_codec_decode().
Signed-off-by: James Almer <jamrial@gmail.com>
Some callers (like do_subtitle_out(), or do_streamcopy()) call this
with an AVPacket that is not refcounted. This can cause undefined
behavior.
Calling av_packet_move_ref() does not make a packet refcounted if it
isn't yet. (And it can't be made to, because it always succeeds,
and can't return ENOMEM.)
Call av_packet_ref() instead to make sure it's refcounted.
Cc: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Add tests for upmixing and downmixing with audio channel counts that
have a corresponding default layout and also tests where there is no
default layout.
Update the existing "stereo4" test so it actually outputs stereo like
the other stereo tests. Rename the previous "stereo4" test into
"upmix1".
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Also remove the superfluous aandcttables dependency from all the modules
that only need it because of mpegvideoenc
Fixes ticket #7333
Signed-off-by: James Almer <jamrial@gmail.com>
This can change the the MSS value announced to the other end in
the initial TCP packet, it's can be used when failed Path MTU
discovery.
Signed-off-by: Jun Zhao <mypopydev@gmail.com>
Current implementations of qsv components incorrectly work with async level, they
actually try to work in async+1 level stepping into MFX_WRN_DEVICE_BUSY and polling
loop. This change address this misbehaviour.
Signed-off-by: Dmitry Rogozhkin <dmitry.v.rogozhkin@intel.com>
Cc: Maxym Dmytrychenko <maxim.d33@gmail.com>
Cc: Zhong Li <zhong.li@intel.com>
Signed-off-by: Maxym Dmytrychenko <maxim.d33@gmail.com>
Rematrixing supports up to 64 channels. However, there is only a limited number of channel layouts defined. Since the in/out channel count is currently obtained from the channel layout, for undefined layouts (e.g. for 9, 10, 11 channels etc.) the rematrixing fails.
This patch changes rematrix init methods to use in (used) and out channel count directly instead of computing it from channel layout.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
libavutil/hwcontext_d3d11va.c: In function 'd3d11va_device_create':
libavutil/hwcontext_d3d11va.c:554:46: warning: passing argument 2 of 'pAdapter->lpVtbl->GetDesc' from incompatible pointer type [-Wincompatible-pointer-types]
hr = IDXGIAdapter2_GetDesc(pAdapter, &desc);
^
libavutil/hwcontext_d3d11va.c:554:46: note: expected 'DXGI_ADAPTER_DESC * {aka struct DXGI_ADAPTER_DESC *}' but argument is of type 'DXGI_ADAPTER_DESC2 * {aka struct DXGI_ADAPTER_DESC2 *}'
Reviewed-by: Jean-Baptiste Kempf <jb@videolan.org>
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes a compilation warning if size_t != uint64_t:
libavformat/mov.c: In function ‘mov_read_saio’:
libavformat/mov.c:6207:45: warning: assignment from incompatible pointer type [-Wincompatible-pointer-types]
encryption_index->auxiliary_offsets = auxiliary_offsets;
^
This way if an index table segment is present multiple times, we can always use
the proper one instead of the invalid one.
Fixes seeking in the sample of ticket #5671.
Signed-off-by: Marton Balint <cus@passwd.hu>
And add it to the CONFIGURABLE_COMPONENTS list in Makefile. This way, changes
to the new file will be tracked and the usual warning to suggest re-running
configure will be shown.
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes the following warnings:
In file included from libavcodec/fft_fixed.c:21:0:
libavcodec/fft_template.c:528:6: warning: ‘pass_big’ defined but not used [-Wunused-function]
PASS(pass_big)
^
libavcodec/fft_template.c:505:13: note: in definition of macro ‘PASS’
static void name(FFTComplex *z, const FFTSample *wre, unsigned int n)\
^~~~
CC libavcodec/ffv1.o
In file included from libavcodec/fft_float.c:21:0:
libavcodec/fft_template.c:528:6: warning: ‘pass_big’ defined but not used [-Wunused-function]
PASS(pass_big)
^
libavcodec/fft_template.c:505:13: note: in definition of macro ‘PASS’
static void name(FFTComplex *z, const FFTSample *wre, unsigned int n)\
^~~~
GCC requires the argument to vec_splat_u32 to be a literal. The easiest
way to accomplish this is to change 'shift' to be const in scale (as it
is in the transform routine above), and convert both routines to be
inline. This way, GCC can coerce the values to literals.
Tested on a 970 (Apple G5) and POWER9 (Talos II); passed fate and played
a clip of Big Buck Bunny correctly.
Fixes ticket #7048
Signed-off-by: A. Wilcox <AWilcox@Wilcox-Tech.com>
aom_codec_get_global_headers() is not implemented as of libaom 1.0.0 for AV1, so
we're forced to extract the relevant header OBUs from the first packet and propagate
them as packet side data.
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes two warnings:
libavfilter/af_afir.c:194:45: warning: assuming signed overflow does not occur when assuming that (X - c) > X is always false [-Wstrict-overflow]
int dx = FFABS(x1-x0), sx = x0 < x1 ? 1 : -1;
~~~~~~~~~~~~^~~~
libavfilter/af_aiir.c:689:45: warning: assuming signed overflow does not occur when assuming that (X - c) > X is always false [-Wstrict-overflow]
int dx = FFABS(x1-x0), sx = x0 < x1 ? 1 : -1;
~~~~~~~~~~~~^~~~
On macOS, a zero rc_max_rate cause an error from
VTSessionSetProperty(kVTCompressionPropertyKey_DataRateLimits).
on iOS (depending on device/version), a zero rc_max_rate cause invalid
arguments from the vtenc_output_callback after few frames and then a crash
within the VideoToolbox library.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Fixes a bug that would prevent using multiple comma-separated filters,
and allows options to be passed to each filter.
Based on similar loop in ffmpeg_opt.c's new_output_stream().
Signed-off-by: Aman Gupta <aman@tmm1.net>
Lensfun is a library that applies lens correction to an image using a
database of cameras/lenses (you provide the camera and lens models, and
it uses the corresponding database entry's parameters to apply lens
correction). It is licensed under LGPL3.
The lensfun filter utilizes the lensfun library to apply lens
correction to videos as well as images.
This filter was created out of necessity since I wanted to apply lens
correction to a video and the lenscorrection filter did not work for me.
While this filter requires little info from the user to apply lens
correction, the flaw is that lensfun is intended to be used on indvidual
images. When used on a video, the parameters such as focal length is
constant, so lens correction may fail on videos where the camera's focal
length changes (zooming in or out via zoom lens). To use this filter
correctly on videos where such parameters change, timeline editing may
be used since this filter supports it.
Note that valgrind shows a small memory leak which is not from this
filter but from the lensfun library (memory is allocated when loading
the lensfun database but it somehow isn't deallocated even during
cleanup; it is briefly created in the init function of the filter, and
destroyed before the init function returns). This may have been fixed by
the latest commit in the lensfun repository; the current latest release
of lensfun is almost 3 years ago.
Bi-Linear interpolation is used by default as lanczos interpolation
shows more artifacts in the corrected image in my tests.
The lanczos interpolation is derived from lenstool's implementation of
lanczos interpolation. Lenstool is an app within the lensfun repository
which is licensed under GPL3.
v2 of this patch fixes license notice in libavfilter/vf_lensfun.c
v3 of this patch fixes code style and dependency to gplv3 (thanks to
Paul B Mahol for pointing out the mentioned issues).
v4 of this patch fixes more code style issues that were missed in
v3.
v5 of this patch adds line breaks to some of the documentation in
doc/filters.texi (thanks to Gyan Doshi for pointing out the issue).
v6 of this patch fixes more problems (thanks to Moritz Barsnick for
pointing them out).
v7 of this patch fixes use of sqrt() (changed to sqrtf(); thanks to
Moritz Barsnick for pointing this out). Also should be rebased off of
latest master branch commits at this point.
Signed-off-by: Stephen Seo <seo.disparate@gmail.com>
MP3 frames may not be aligned to aa chunk boundaries. When seeking,
calculate the expected frame offset in the target chunk. Adjust the
timestamp and truncate the next packet accordingly.
This solution works for the majority of tested audio material. For
some rare encodings with mp3 padding or embedded id3 tags, it will
mispredict the correct offset, and at worst skip an extra frame.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In commit 975a1a8,function ff_vc1_h_s_overlap_mmi was refactored,
but the declaration in libavcodec/mips/vc1dsp_mips.h was unchanged.
Change-Id: I90beae683511622a0cc1130ab1660ac8669ec3ef
Signed-off-by: Shiyou Yin <yinshiyou-hf@loongson.cn>
Reviewed-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Without this check some crafted files might crash because a packet might be
demuxed which have no corresponding mxf track.
Signed-off-by: Marton Balint <cus@passwd.hu>
This was reduced from 128 in libav commit
192f1984b1, but since we support unknown channel
layouts, we can increase this limit.
Fixes ticket #6332.
Signed-off-by: Marton Balint <cus@passwd.hu>
SDL from version 2.0.8 has support for full range YUV and specifying
BT601/BT709 color space for YUV->RGB conversion.
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes the following warnings:
libavcodec/aarch64/h264dsp_init_aarch64.c: In function ‘ff_h264dsp_init_aarch64’:
libavcodec/aarch64/h264dsp_init_aarch64.c:84:38: warning: assignment from incompatible pointer type [enabled by default]
c->weight_h264_pixels_tab[0] = ff_weight_h264_pixels_16_neon;
^
libavcodec/aarch64/h264dsp_init_aarch64.c:85:38: warning: assignment from incompatible pointer type [enabled by default]
c->weight_h264_pixels_tab[1] = ff_weight_h264_pixels_8_neon;
^
libavcodec/aarch64/h264dsp_init_aarch64.c:86:38: warning: assignment from incompatible pointer type [enabled by default]
c->weight_h264_pixels_tab[2] = ff_weight_h264_pixels_4_neon;
^
libavcodec/aarch64/h264dsp_init_aarch64.c:88:40: warning: assignment from incompatible pointer type [enabled by default]
c->biweight_h264_pixels_tab[0] = ff_biweight_h264_pixels_16_neon;
^
libavcodec/aarch64/h264dsp_init_aarch64.c:89:40: warning: assignment from incompatible pointer type [enabled by default]
c->biweight_h264_pixels_tab[1] = ff_biweight_h264_pixels_8_neon;
^
libavcodec/aarch64/h264dsp_init_aarch64.c:90:40: warning: assignment from incompatible pointer type [enabled by default]
c->biweight_h264_pixels_tab[2] = ff_biweight_h264_pixels_4_neon;
^
At present, box size is clipped to frame size before being drawn,
which can lead to the box not fully covering animated text which is
longer than one or both frame dimensions.
Since ff_blend_rectangle correctly takes care of clipping, it is skipped
here which results in correct box sizing
This fixes the check for the reserved MPEG audio version ID,
used to detect an invalid frame header.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
fix ticket: 7305
vs->sequence - hls->start_sequence - vs->nb_entries is the
after_init_list_dur fragment numbers
fix the wrong compute way vs->sequence - vs->nb_entries
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Set make variable KEEP to non-zero value to preserve temp files
when a test has passed.
Helpful in diagnosing failed tests when test outfile is some type of
single hash and does not reveal differences in processed output.
The version 1 needs the channel count and would divide by 0
Fixes: division by 0
Fixes: fpe_movenc.c_1108_1.ogg
Fixes: fpe_movenc.c_1108_2.ogg
Fixes: fpe_movenc.c_1108_3.wav
Found-by: #CHEN HONGXU# <HCHEN017@e.ntu.edu.sg>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
read_packet reads content in chunks. Thus seek must be clamped to valid
chunk positions in the file, which in turn are relative to chapter start
positions.
So in read_header, scan for chapter headers once by skipping through the
content. Set stream time_base based on bitrate in bytes/s, for easy
timestamp to position conversion.
Then in read_seek, find the chapter containing the seek position, calculate
the nearest chunk position, and reinit the read_seek state accordingly.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Check the MPEG version ID for the reserved bit pattern 01, and abort the
header check in that case. This reduces the chance of misinterpreting
arbitrary data as a valid header, and prevents resulting audio artifacts.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Remember the end position of audio content in the file and check it during
read_packet. There always seems to be other data beyond it, which could be
misinterpreted as more audio. Also add some extra avio_read error checks,
to bail early in case of a broken/truncated file.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is identical to what the VP9 parser does
Fixes: 9215/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_LIBVPX_VP8_fuzzer-5768227253649408
Fixes: out of memory access
This may also fix oss fuzz issue 9212
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: use after free()
Fixes: rmdec-crash-ffe85b4cab1597d1cfea6955705e53f1f5c8a362
Found-by: Paul Ch <paulcher@icloud.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: integer overflow and out of array access
Fixes: asfo-crash-46080c4341572a7137a162331af77f6ded45cbd7
Found-by: Paul Ch <paulcher@icloud.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: out of array access
Fixes: mxf-crash-1c2e59bf07a34675bfb3ada5e1ec22fa9f38f923
Found-by: Paul Ch <paulcher@icloud.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: out of array read
Fixes: asff-crash-0e53d0dc491dfdd507530b66562812fbd4c36678
Found-by: Paul Ch <paulcher@icloud.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
For clip wrapped essences this should work. Also, since index_edit_rate can now
be different from track edit rate, remove overriding track edit rate.
Signed-off-by: Marton Balint <cus@passwd.hu>
The profile field is changed by code inside and outside the decoder,
its not a reliable indicator of the internal codec state.
Maintaining it consistency with studio_profile is messy.
Its easier to just avoid it and use only studio_profile
Fixes: assertion failure
Fixes: ffmpeg_crash_9.avi
Found-by: Thuan Pham, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 1139785606 + 1454196085 cannot be represented in type 'int'
Fixes: 8937/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-6202943597445120
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This exposes encryption info from the container to the app. This
includes key ID, IV, and subsample byte ranges. The info is passed
using the new side-data AV_PKT_DATA_ENCRYPTION_DATA and
AV_PKT_DATA_ENCRYPTION_INIT_DATA.
Signed-off-by: Jacob Trimble <modmaker@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit implements a full ATRAC9 decoder, a simple low-delay codec
developed by Sony and used in most PSVita games, some PS3 games and some
PS4 games. Its similar to AAC in that it uses Huffman coded scalefactors
but instead of vector quantization it just Huffman codes the spectral
coefficients (in a way similar to how Opus splits band energy coding
into coarse and fine precision). It opts to write rather large Huffman
codes by packing several small coefficients into one Huffman coded
symbol, though I don't believe this increases efficiency at all.
Band extension implements SBC in a simple way, first it mirrors the
lower spectrum onto the higher frequencies and then it uses one of 5
filters to shape it. Noise substitution is implemented via 2 of them.
Unlike previous ATRAC codecs, there's no QMF, this is a standard MDCT
codec.
Based off of the reverse engineering work of Alex Barney.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Testcase with large transparent rectangles changes from 67 sec to 3 sec decode time
Fixes: Timeout
Fixes: 8728/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DVDSUB_fuzzer-5190088756559872
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
And let the generic code figure out the proper PTS. This is needed because apng
does not provide seek functions, but after a generic seek (e.g. to file start)
timestamps are not reset which causes broken timestamps when looping apngs,
like in ticket #6121.
Signed-off-by: Marton Balint <cus@passwd.hu>
The input thread needs to be properly cleaned up and re-initalized before we
can start reading again in threaded mode. (Threaded input reading is used when
there is mode than one input file).
Fixes ticket #6121 and #7043.
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes: out of array access
Fixes: ffmpeg_bof_4.avi
Fixes: ffmpeg_bof_5.avi
Fixes: ffmpeg_bof_6.avi
Found-by: Thuan Pham, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Reviewed-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Set pixel format and color_range for YUVJ pixel formats. Also set
color_range based on AVFormatContext.
Signed-off-by: Wang Cao <wangcao@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The overlap filter is not correct for vertical edges in frame interlaced
I and P pictures. When filtering macroblocks with different FIELDTX values,
we have to match the lines at both sides of the vertical border. In addition,
we have to use the correct rounding values, depending on the line we are
filtering.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
The overlap filter needs to cover a full macroblock vertical edge when the
FIELDTX value for two neighbouring macroblocks is not equal. By changing
the internal ordering of the blocks from row major to column major, we do
not need to reinterlace a FIELDTX coded macroblock before running the overlap
filter.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
When not using libavformat for demuxing, AVCodecContext.has_b_frames
gets set too late causing the recovery frame heuristic in h264_refs to
incorrectly flag an early frame as recovered.
This patch sets has_b_frames earlier to prevent improperly flagging the
frame as recovered.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: null pointer dereference
Fixes: ffmpeg_crash_7.avi
Found-by: Thuan Pham, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: out of array read
Fixes: ffmpeg_crash_8.avi
Found-by: Thuan Pham, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: null pointer dereference
Fixes: ffmpeg_crash_6.avi
Found-by: Thuan Pham, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: out of array access
Fixes: ffmpeg_bof_1.avi
Found-by: Thuan Pham, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -4096 * -524288 cannot be represented in type 'int'
Fixes: 8650/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RA_144_fuzzer-5734816036159488
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes a bug/regression with very small packets
Fixes: output_file
Regression since: 0782fb6bcb
Reported-by: Thierry Foucu <tfoucu@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Also make sure we set a valid track index sid and a valid track edit rate in
order for the index to be useful.
Signed-off-by: Marton Balint <cus@passwd.hu>
It is possible for there to be multiple encryption init info structure.
For example, to support multiple key systems or in key rotation. This
changes the AVEncryptionInitInfo struct to be a linked list so there
can be multiple structs without breaking ABI.
Signed-off-by: Jacob Trimble <modmaker@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This filter does HDR(HDR10/HLG) to SDR conversion with tone-mapping.
An example command to use this filter with vaapi codecs:
FFMPEG -init_hw_device vaapi=va:/dev/dri/renderD128 -init_hw_device \
opencl=ocl@va -hwaccel vaapi -hwaccel_device va -hwaccel_output_format \
vaapi -i INPUT -filter_hw_device ocl -filter_complex \
'[0:v]hwmap,tonemap_opencl=t=bt2020:tonemap=linear:format=p010[x1]; \
[x1]hwmap=derive_device=vaapi:reverse=1' -c:v hevc_vaapi -profile 2 OUTPUT
Signed-off-by: Ruiling Song <ruiling.song@intel.com>
Current ffplay code assumes that the read thread is in its main loop before any
key events are captured, but apparently on IOS even keypresses without a window
are forwared.
Fixes ticket #7252.
Signed-off-by: Marton Balint <cus@passwd.hu>
da9cc22d5b allowed the MOV muxer to relay a custom stream handler name,
whether populated from the input stream or user-set. However, the entry
key didn't match the key set by the MOV demuxer, so it wasn't
effective. Fixed.
Due to the change, four FATE refs have to be updated. Verified that the
target payload of the tests hasn't changed in terms of CRC.
CLI options -maxrate, -bufsize and -rc_init_occupancy can now be picked
up by the x265 wrapper. Min. rc init has to be 1001 to avoid x265
setting it to vbv-bufsize.
In 9152c1e495, the mpegts parser was taught how to parse
PMT sections which contained multiple tables. That commit
fixed parsing of PMT packets from some cable providers,
which included a special SCTE table (0xc0) before the
standard program map table (0x2).
Sometimes, however, the combined 0xc0 and 0x2 tables are
larger than a single TS packet (188 bytes). The mpegts parser
already attempts to parse sections which span multiple packets,
but still assumed that the split section only contained one
table.
This patch fixes parsing of such a sample[1].
Before:
Input #0, mpegts, from 'combined-pmt-tids-split.ts':
Duration: 00:00:01.26, start: 39188.931756, bitrate: 597 kb/s
Program 1
No Program
Stream #0:0[0xeff]: Audio: ac3, 48000 Hz, mono, fltp, 64 kb/s
Stream #0:1[0xefd]: Audio: mp3, 0 channels, fltp
Stream #0:2[0xefe]: Unknown: none
After:
Input #0, mpegts, from 'combined-pmt-tids-split.ts':
Duration: 00:00:01.27, start: 39188.931756, bitrate: 589 kb/s
Program 1
Stream #0:0[0xefd]: Video: h264 ([27][0][0][0] / 0x001B), none, 59.94 fps, 59.94 tbr, 90k tbn, 180k tbc
Stream #0:1[0xefe](eng): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, stereo, fltp, 384 kb/s
Stream #0:2[0xeff](spa): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, mono, fltp, 64 kb/s
Stream #0:3[0xf00]: Data: scte_35
Stream #0:4[0xf01]: Unknown: none (ETV1 / 0x31565445)
Stream #0:5[0xf02]: Unknown: none (ETV1 / 0x31565445)
Stream #0:6[0xf03]: Unknown: none ([192][0][0][0] / 0x00C0)
With the patch, the PMT is parsed correctly so the streams are
created in the correct order, are associated with "Program 1",
and their codecs are set correctly.
[1] https://s3.amazonaws.com/tmm1/combined-pmt-tids-split.ts
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
@xyz reported a regression on his Sony Xperia Z3 Tablet Compact where
playback would intermittently fail to start, essentially deadlocking in
the decoder. Bisecting narrowed down the issue to this commit, which was
meant as an optimization but is not necessary.
This reverts commit a75bb5496a.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Split vtenc_init() into vtenc_init() (VTEncContext initialization) and
vtenc_configure_encoder() (creates the vt session).
This commit will allow to restart the vt session while encoding.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Simple and Main Profile also need unsigned put_pixels_clamped. Add an argument
to choose between signed and unsigned put_pixels and change function name to
vc1_put_blocks_clamped.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
When using hardware accelerated decoding for multi-slice field interlaced pictures,
only the first slice was decoded. This patch adds the neccesary looping over the
remaining slices that may exist in field interlaced pictures. Additionally, we align
the calculation of mby_start for the second field with the method given in VC-1 spec.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Contrary to VC-1 spec, VAAPI expects the row address of the first
macroblock row in the first slice to start from zero for the second
field in a field interlaced picture.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Simple and Main profile also need unsigned put_pixels_clamped. Add an argument
to choose between signed and unsigned put_pixels and change function name to
vc1_put_blocks_clamped.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Overlap filtering I and BI frames for Simple and Main profile is only
dependent on PQUANT. Restrict testing for CONDOVER and OVERFLAGS to
advanced profile. Change from mb_width to end_mb_x in ff_vc1_i_loop_filter
to avoid breaking the Microsoft Screen 2 decoder.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
The v_overlap_filter needs to run on the colocated block of the previous
macroblock. For the luma plane, the colocated block is located two blocks
on the left instead of one. In addition, the overlap filter needs to run
on the non-edge blocks of the first macroblock row and column.
Fixes ticket #7171.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Fixes: signed integer overflow: 1195517 * 2048 cannot be represented in type 'int'
Fixes: 8636/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-4695836326887424
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: negation of -2147483648 cannot be represented in type 'int'; cast to an unsigned type to negate this value to itself
Fixes: 8521/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DIRAC_fuzzer-5639024952737792
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
STRIDE_ALIGN is not known in libavutil so av_image_check_size* cannot consider it
Fixes: OOM
Fixes: 8291/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SNOW_fuzzer-5176528009691136
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Regression since: af1761f7
Fixes: Division by 0
Fixes: ffmpeg_crash_1
Found-by: Thuan Pham, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
verify that the stco atom is upgraded to co64 when the addition of moov
size to the offsets results in an overflow
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
when the last offsets in the stco atom are close to 4GB, the addition of
the moov atom size can overflow, causing corruption near the end of the
mp4 file.
this patch upgrades all stco atoms to co64 when such an edge case is
detected. in order to accomplish this, the implementation was changed to
walk the atom tree, instead of searching for the strings 'stco'/'co64'.
this was required since when an stco atom is changed to co64, its size
changes, and the sizes of all containing atoms (moov, trak, etc.) have
to be updated as well.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: assertion failure
Fixes: ffmpeg_crash_5.avi
Found-by: Thuan Pham <thuanpv@comp.nus.edu.sg>, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
For chapter images, the mov demux produces streams with disposition set
to attached_pic+timed_thumbnails. This patch fixes to properly recognize
streams that should be encoded as cover image (ones with only and only
attached_pic disposition set).
Signed-off-by: Timo Teräs <timo.teras@iki.fi>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Find codec tag for attached images using appropriate list of
supported image formats.
This fixes writing the cover image to m4v/m4a and other container
formats that do not allow these codecs as a track.
Signed-off-by: Timo Teräs <timo.teras@iki.fi>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
A generic lavf flag for AAC LATM packetization for the RTP muxer was
added in ef409645f0 and then made inert 20 days later in 0832122880
when a private muxer option was added and the generic flag no longer
read.
If the user provides a valid timecode_format look for timecode of that
format in the capture and if found store it on the video avstream's
metadata.
Slightly modified by Marton Balint to capture per-frame timecode as well.
Signed-off-by: Marton Balint <cus@passwd.hu>
The default memory allocator is limited in the max number of frames available,
and therefore caused frame drops if the frames were not freed fast enough.
Signed-off-by: Marton Balint <cus@passwd.hu>
Some of these enums have gaps in between their values, since they correspond
to the values in various specs, instead of being an incrementing list.
Fixes segfaults when, for example, using the valid API call:
av_color_primaries_from_name("jecdec-p22");
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
1. vcm mode is only available for H264.
2. vcm is not supported on Linux, but it is shown when run "./avconv -h
encoder=h264_qsv |grep vcm". This shouldn't happen.
Signed-off-by: Zhong Li <zhong.li@intel.com>
Signed-off-by: Maxym Dmytrychenko <maxim.d33@gmail.com>
Use a common way to control target_usage, keeping consistent with vaapi
encoders. The private option preset is kept only for compatibility.
Signed-off-by: Zhong Li <zhong.li@intel.com>
Signed-off-by: Maxym Dmytrychenko <maxim.d33@gmail.com>
Fixes: signed integer overflow: 2146907204 + 26846088 cannot be represented in type 'int'
Fixes: 8105/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WAVPACK_fuzzer-6233036682166272
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes the following warnings:
In file included from libswscale/rgb2rgb.c:128:0:
libswscale/rgb2rgb_template.c:346:13: warning: 'shuffle_bytes_3210_c' defined but not used
libswscale/rgb2rgb_template.c:346:13: warning: 'shuffle_bytes_3012_c' defined but not used
libswscale/rgb2rgb_template.c:346:13: warning: 'shuffle_bytes_1230_c' defined but not used
move the the function aacsbr_tableinit definition from header file
to .c file to fix make checkheaders warning.
Signed-off-by: Jun Zhao <mypopydev@gmail.com>
move the the function init_tables() definitions from header file
to .c file to fix make checkheaders warning.
Signed-off-by: Jun Zhao <mypopydev@gmail.com>
This validates that the common encryption saio/saiz atoms only appear
when the data is actually encrypted. This also ignores those atoms
in clear content.
Found by Chrome's ClusterFuzz: http://crbug.com/850389
Signed-off-by: Jacob Trimble <modmaker@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 3 * 1006632960 cannot be represented in type 'int'
Fixes: 8278/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-5692857166856192
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: OOM
Fixes: 8195/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-5179785826271232
The reference software appears to use longs for 32bits and it uses int for nmeans
hinting that the intended maximum size was not 32bit.
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fix a build error on Windows:
C2440: connot convert from 'void (__cdecl *) (...)' to 'void (__stdcall *)(...)'.
Signed-off-by: Ruiling Song <ruiling.song@intel.com>
Fixes: signed integer overflow: 2147483647 + 1 cannot be represented in type 'int'
Fixes: 8024/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-5109204648984576
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 32768 + 2147450880 cannot be represented in type 'int'
Fixes: 7885/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_THP_fuzzer-5298834394578944
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 1077952576 + 1077952576 cannot be represented in type 'int'
Fixes: 7712/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TRUEMOTION2_fuzzer-5056281753681920
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If start_time is not set, ffmpeg takes the duration from the global
movie instead of the per stream duration.
Signed-off-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
We already do this for audio, but it should be done for video too.
If we don't, seeking back to the start of the file, for example, can
become quite broken, since the first N packets will have repeating
and nonmonotonic PTS, yet they need to be decoded even if they are
to be discarded.
Signed-off-by: Sasi Inguva <isasi@isasi.mtv.corp.google.com>
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Fixes: runtime error: signed integer overflow: -1440457022 - 785819492 cannot be represented in type 'int'
Fixes: 7700/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_OPUS_fuzzer-6595838684954624
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Forced key frames generation functionality was assuming the first PTS
value as zero, but, when 'copyts' is enabled, the first PTS can be any
big number. This was eventually forcing all the frames as key frames.
To resolve this issue, update has been made to use first input pts as
reference pts.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Out-of-bounds reference pixel replication should take into account the frame
coding mode of the reference frame(s), not the frame coding mode of the
current frame.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
HMS is formatted as HH:MM:SS.mmm, but, HH part is not limited to
24 hours. For example, the the drawn text may look like this:
243029:20:30.342. To present the timestamp in more readable and
user friendly format, this patch provides an additional option
to limit the hour part in the range 0-23.
Note: Actually the above required format can be obtained with
format options 'localtime' and 'gmtime', but, milliseconds part
is not supported in those formats.
These files depend on libavformat, and the vf_srcnn filter
currently is the only thing utilizing these dnn_* files and
already happens to have a dependency on libavformat.
This fixes compilation in cases where libavformat is not a
dependency for libavfilter.
Reported by Kam_ on IRC.
These 2 fields are not always the same, it is simpler to always use the same field
for detecting studio profile
Fixes: null pointer dereference
Fixes: ffmpeg_crash_3.avi
Found-by: Thuan Pham <thuanpv@comp.nus.edu.sg>, Marcel Böhme, Andrew Santosa and Alexandru RazvanCaciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Generic C implementation of vf_blend performs reads and writes of 16-bit
elements, which requires the buffers to be aligned to at least 2-byte
boundary.
Also, the change fixes source buffer overrun caused by src_offset being
added to to test handling of misaligned buffers.
Fixes: #7226
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
1. validate the moov size before checking for cmov atom
2. avoid performing arithmetic operations on unvalidated numbers
3. verify the stco/co64 offset count does not overflow the stco/co64
atom (not only the moov atom)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This improves performance and makes qtrle behave more similar to other decoders.
Libavcodec does generally not output known duplicated frames, instead the calling Application
can insert them as it needs.
Fixes: Timeout
Fixes: 6383/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_QTRLE_fuzzer-6199846902956032
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The producer reference time box supplies relative wall-clock times
at which movie fragments, or files containing movie fragments
(such as segments) were produced.
The box is mainly useful in live streaming use cases. A media player
can parse the box and utilize the time fields to measure and improve
the latency during real time playout.
This utility function creates 64-bit NTP time format as per the RFC
5905.
A simple explaination of 64-bit NTP time format is here
http://www.beaglesoft.com/Manual/page53.htm
Direct prediction for interlace frame B pictures references the mv in the
second block in an MB in the backward reference frame for the twomv case.
When the backward reference frame is an I frame, this value may be unset.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
For interlace field pictures s->mb_height indicates the height of the full
picture in MBs, i.e. the two fields combined. A single field is half this
size. When calculating mquant for interlace field pictures, the bottom edge
is the last MB row of the field.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
The last workaround is not sufficient to make oss fuzz work with the iterate API
as it did not provide a FFmpeg that external libs can be linked to.
This patch does not fully restore the pre iterate functionality. My attempts to
do this have so far failed.
The problem with this solution is that it renders the fuzzers virtual system
ffmpeg (libs) non functional. Which differs from a real system compared to the
virtual system tested by the fuzzer.
It should theoretically not matter as the system ffmpeg wouldnt be used.
But with more cases being fuzzed we likely will hit a case where a external
lib is involved and it does matter ...
Working around this may be possible with weak symbols but so far my attempts
failed
Alternatively multiple ffmpeg could be built, this becomes messy though
quickly as they need to be all linked together. That is we need a FFmpeg
that has the iterate API modified so it can work with the resources
available to ossfuzz. And at the same time we need a ffmpeg that has
its full functionality for any external libs which use ffmpeg and are
used by ffmpeg.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Yet another case of forgotten 0 =! EOF translation.
While the documentation for this specific synchronous read
function does not mention it, the documentation for
`sftp_async_read` documents it, as well as looking at the
implementation of this function leads one to find
`if (handle->eof) { return 0; }`.
Reported by stnutt on IRC.
Applicable only to webm output format.
By default all the segment filenames end with .m4s extension.
When someone chooses webm output format, we recommend they also override the relevant segment name options to end with .webm extension. This patch will issue a warning for he same
Right now segment file format is chosen to be either mp4 or webm based on the codec format.
This patch makes that choice configurable by the user, instead of being decided by the muxer.
Also with this change per-stream choice segment file format(based on codec type) is not possible.
All the output audio and video streams should be in the same file format.
Fixes: left shift of 1 by 63 places cannot be represented in type 'long long'
Fixes: out of array access
Fixes: 7284/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AC3_fuzzer-5767914968842240
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -2141499320 + -14469590 cannot be represented in type 'int'
Fixes: 7351/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-6351214791884800
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 1073741842 + 1784008138 cannot be represented in type 'int'
Fixes: 6792/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-5677589835284480
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes truncation
Fixes Assertion n <= 31 && value < (1U << n) failed at libavcodec/put_bits.h:169
Fixes: ffmpeg_crash_2.avi
Found-by: Thuan Pham <thuanpv@comp.nus.edu.sg>, Marcel Böhme, Andrew Santosa and Alexandru RazvanCaciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: inconsistency
Fixes:runtime error: index 8 out of bounds for type 'int32_t [8]'
Fixes: 6686/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TRUEHD_fuzzer-5191383498358784
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This avoids inconsistent value combinations.
Alternatively it would be possible to add more checks and careful use of
temporary variables, but my try of this quickly seemed to become
a rather large change.
The disadvantage of this, is that the struct is copied back and forth.
Fixes: index 6 out of bounds for type 'const uint16_t [5][16]'
Fixes: 6557/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_INDEO4_fuzzer-4787296550256640
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Provide proper aliases to enable/disable MFE.
The numeric values are ambiguous and misleading (e.g: user may misunderstand
setting mfmode to 1 is to enable MFE but actually it is to disable MFE, and
set it to be 5 or above is meaningless).
MFX_MF_MANUAL hasn't been exposed since it is to be implemented.
Signed-off-by: Zhong Li <zhong.li@intel.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The libvpx doxy says that a value of 0 for the g_threads field is
equivalent to a value of 1, whereas for avctx->thread_count it means
the maximum amount of threads possible for the host system.
Use av_cpu_count() to get the correct thread count when auto threads
is requested.
Reviewed-by: James Zern <jzern@google.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This uses any devices it can find on the host system - on a system with no
hardware device support or in builds with no support included it will do
nothing and pass.
Both stream_id and stream_identifier are used in this file,
and have different meanings. The latter comes from the
stream_identifier_descriptor.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Without this some operations might overflow (undefined behavior)
even though the index adding loop would never execute
No testcase known
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If v->fieldtx_is_raw is not reset to zero, it may spill over from a previous
interlaced frame I/BI picture.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This massively reduces the detection of random data as low score mp3
It may improve security by making it harder to read non multimedia data
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Only for the last slice of the first field is the last line of the slice
equal to the height of the field.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
DIRECTBIT was decoded before the intra/inter MB branching when decoding
interlace frame B pictures. Resulting in mistakenly also decoding it for intra
MBs where this syntax element is not present.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This new optional flag makes it easier to deal with mpegts
samples where the PMT is updated and elementary streams move
to different PIDs in the middle of playback.
Previously, new AVStreams were created per PID, and it was up
to the user to figure out which streams had migrated to a new PID
(by iterating over the list of AVProgram and making guesses), and
switch seamlessly to the new AVStream during playback.
Transcoding or remuxing these streams with ffmpeg on the CLI was
also quite painful, and the user would need to extract each set
of PIDs into a separate file and then stitch them back together.
With this new option, the mpegts demuxer will automatically detect
PMT changes and feed data from the new PID to the original AVStream
that was created for the orignal PID. For mpegts samples with
stream_identifier_descriptor available, the unique ID is used to
merge PIDs together. If the stream id is not available, the demuxer
attempts to map PIDs based on their position within the PMT.
With this change, I am able to playback and transcode/remux these
two samples which previously caused issues:
https://tmm1.s3.amazonaws.com/pmt-version-change.tshttps://kuroko.fushizen.eu/videos/pid_switch_sample.ts
I also have another longer sample in which the PMT changes
repeatedly and ES streams move to different pids three times
during playback:
https://tmm1.s3.amazonaws.com/multiple-pmt-change.ts
Demuxing this sample with the new option shows several new log
messages as the PMT changes are handled:
[mpegts] detected PMT change (program=1, version=3/6, pcr_pid=0xf98/0xfb7)
[mpegts] re-using existing video stream 0 (pid=0xf98) for new pid=0xfb7
[mpegts] re-using existing audio stream 1 (pid=0xf99) for new pid=0xfb8
[mpegts] re-using existing audio stream 2 (pid=0xf9a) for new pid=0xfb9
[mpegts] detected PMT change (program=1, version=6/3, pcr_pid=0xfb7/0xf98)
[mpegts] detected PMT change (program=1, version=3/4, pcr_pid=0xf98/0xf9b)
[mpegts] re-using existing video stream 0 (pid=0xf98) for new pid=0xf9b
[mpegts] re-using existing audio stream 1 (pid=0xf99) for new pid=0xf9c
[mpegts] re-using existing audio stream 2 (pid=0xf9a) for new pid=0xf9d
[mpegts] detected PMT change (program=1, version=4/5, pcr_pid=0xf9b/0xfa9)
[mpegts] re-using existing video stream 0 (pid=0xf98) for new pid=0xfa9
[mpegts] re-using existing audio stream 1 (pid=0xf99) for new pid=0xfaa
[mpegts] re-using existing audio stream 2 (pid=0xf9a) for new pid=0xfab
[mpegts] detected PMT change (program=1, version=5/6, pcr_pid=0xfa9/0xfb7)
Signed-off-by: Aman Gupta <aman@tmm1.net>
With these fields, the user has enough information to
detect PMT changes and switch to new streams when the PMT
is updated with new ES pids.
To do so, the user would monitor the AVProgram they're interested
in for changes to pmt_version. If the version changes, they would
iterate over the program's streams to find new streams added with
the updated version number.
If new versions of streams are found, then the user would first try
to replace existing streams where stream_identifier matched.
If stream_identifier is not available, then the user would compare
pmt_stream_idx instead to replace the stream that was previously
at the same position within the PMT.
Signed-off-by: Aman Gupta <aman@tmm1.net>
These fields will allow the mpegts demuxer to expose details about
the PMT/program which created the AVProgram and its AVStreams.
In mpegts, a PMT which advertises streams has a version number
which can be incremented at any time. When the version changes,
the pids which correspond to each of it's streams can also change.
Since ffmpeg creates a new AVStream per pid by default, an API user
needs the ability to (a) detect when the PMT changed, and (b) tell
which AVStream were added to replace earlier streams.
This has been a long-standing issue with ffmpeg's handling of mpegts
streams with PMT changes, and I found two related patches in the wild
that attempt to solve the same problem:
The first is in MythTV's ffmpeg fork, where they added a
void (*streams_changed)(void*); to AVFormatContext and call it from
their fork of the mpegts demuxer whenever the PMT changes.
The second was proposed by XBMC in
https://ffmpeg.org/pipermail/ffmpeg-devel/2012-December/135036.html,
where they created a new AVMEDIA_TYPE_DATA stream with id=0 and
attempted to send packets to it whenever the PMT changed.
Signed-off-by: Aman Gupta <aman@tmm1.net>
In a normal hwaccel, the AVHWFramesContext sets AVFrame.hw_frames_ctx
when it initializes a new AVFrame in av_hwframe_get_buffer().
But the VT hwaccel doesn't know what hw_frames_ctx to assign when
the AVFrame is first created, because it depends on the format of
the pixbuf that the decoder eventually decides to return. Thus
newly created AVFrames always have a NULL hw_frames_ctx, and the
hwaccel would only assign the ctx once a frame was done decoding.
This worked fine with the H264 decoder, but with the HEVC decoder
the frame's data may be moved to another empty AVFrame. Since the
empty AVFrame never had hw_frames_ctx set, a frame with a NULL
ctx could be returned to the API user.
This patch works around the issue by moving the derived
hw_frames_ctx from the AVFrame to a new VTHWFrame which now holds
both the CVPixelBufferRef and the AVBuffer. The hw_frames_ctx
is only copied to the AVFrame right before it is about to be
returned to the user in videotoolbox_postproc_frame() (since
in the case of VT, the hw_frames_ctx is only there for the API
user anyway).
Fixes playback on macOS and iOS of some hevc videos like
https://s3.amazonaws.com/tmm1/videotoolbox/germany-hevc-zdf.ts
Signed-off-by: Aman Gupta <aman@tmm1.net>
Some filtered mpegts streams may erroneously include PMTs for
programs that are not advertised in the PAT. This confuses ffmpeg
and most players because multiple audio/video streams are created
and it is unclear which ones actually contain data.
See for example https://tmm1.s3.amazonaws.com/unknown-pmts.ts
In this sample, the PAT advertises exactly one program. But the
pid it points to for the program's PMT contains PMTs for other
programs as well. This is because the broadcaster decided to
re-use the same pid for multiple program PMTs.
The hardware that filtered the original multi-program stream
into a single-program stream did so by rewriting the PAT to
contain only the program that was requested. But since it just
passed through the PMT pid referenced in the PAT, multiple PMTs
are still present for the other programs.
Before:
Input #0, mpegts, from 'unknown-pmts.ts':
Duration: 00:00:10.11, start: 80741.189700, bitrate: 9655 kb/s
Program 4
Stream #0:2[0x41]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, bt709, progressive), 1280x720 [SAR 1:1 DAR 16:9], Closed Captions, 11063 kb/s, 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc
Stream #0:3[0x44](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 384 kb/s
Stream #0:4[0x45](spa): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 128 kb/s
No Program
Stream #0:0[0x31]: Video: mpeg2video ([2][0][0][0] / 0x0002), none(tv), 90k tbr, 90k tbn, 90k tbc
Stream #0:1[0x34](eng): Audio: ac3 (AC-3 / 0x332D4341), 0 channels, fltp
Stream #0:5[0x51]: Video: mpeg2video ([2][0][0][0] / 0x0002), none, 90k tbr, 90k tbn
Stream #0:6[0x54](eng): Audio: ac3 (AC-3 / 0x332D4341), 0 channels
With skip_unknown_pmt=1:
Input #0, mpegts, from 'unknown-pmts.ts':
Duration: 00:00:10.11, start: 80741.189700, bitrate: 9655 kb/s
Program 4
Stream #0:0[0x41]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, bt709, progressive), 1280x720 [SAR 1:1 DAR 16:9], Closed Captions, 11063 kb/s, 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc
Stream #0:1[0x44](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 384 kb/s
Stream #0:2[0x45](spa): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 128 kb/s
Signed-off-by: Aman Gupta <aman@tmm1.net>
Generates color bar test patterns based on EBU PAL recommendations.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
The FATE tests for MSVC versions older than 2013 are untested in FATE
and apparently are no longer supported.
This commit makes the configure process error out in case an older version
is used, and suggests to use a supported version of MSVC to compile.
This also changes the documentation to reflect this.
As discussed on IRC:
2018-05-12 19:45:16 jamrial then again, most of those were for old msvc, and i think we're not supporting versions older than 2013 (first one c99 compliant) anymore
2018-05-12 19:45:43 +JEEB yea, I think 2013 update 2 is needed
22:53 <@atomnuker> nevcairiel: which commit broke/unsupported support for msvc 2013?
23:23 <@atomnuker> okay, it was JEEB
23:25 <+JEEB> which was for 2012 and older
23:25 <+JEEB> and IIRC we no longer test those in FATE so that was my assumption
23:26 <+JEEB> 2013 is when MS got trolled enough to actually update their C part
23:26 <+JEEB> aand actually advertised FFmpeg support
23:26 <+JEEB> (although it was semi-failing until VS2013 update 1 or 2)
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Parses the video_stream_descriptor (H.222 2.6.2) to look
for the still_picture_flag. This is exposed to the user
via a new AV_DISPOSITION_STILL_IMAGE.
See for example https://tmm1.s3.amazonaws.com/music-choice.ts,
whose video stream only updates every ~6 seconds.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
For seekable mpegts streams, duration is calculated from
pts by seeking to the end of the file for a pts and subtracting
the initial pts to compute a duration.
This can be expensive in terms of added latency during
probe, especially when streaming over a network. This new
option lets you skip the duration calculation, which is useful
when you don't care about the value and want to save some overhead.
This patch is particularly useful when dealing with live mpegts
streams. Normally such streams are not seekable, so durations
are not calculated. However in my case I am dealing with a seekable
live mpegts stream (networked access to a .ts file which is still
being appended to).
Signed-off-by: Aman Gupta <aman@tmm1.net>
Unbreaks files with unknown extradata, the Canopus decoder accepts both files
with and without this extradata (24 byte "INFO", 16 byte "RDRT", rest "FIEL").
Reported-by: Peter Bubestinger
Tested-by: Piotr Bandurski
Fixes: runtime error: shift exponent -1 is negative
Fixes: 7486/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-4977380939530240
Fixes: runtime error: index 36 out of bounds for type 'const uint8_t [32]'
Fixes: 7566/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-6536620682510336
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: runtime error: left shift of 1876744317 by 16 places cannot be represented in type 'int'
Fixes: 6799/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_G2M_fuzzer-5115274731716608
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
A few days ago ossfuzz stoped testing new FFmpeg as it run out of diskspacee
https://oss-fuzz-build-logs.storage.googleapis.com/index.html
An alternative would be to revert the API.
This changes for example
-rwxr-x--- 1 michael michael 144803654 May 14 12:54 tools/target_dec_ac3_fixed_fuzzer*
to
-rwxr-x--- 1 michael michael 30333852 May 14 12:51 tools/target_dec_ac3_fixed_fuzzer*
Which should massively decrease space requirements
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
My conversation from AVFormatContext->filename to AVFormatContext->url was
wrong in this case because get_chunk_filename uses filename as an output
buffer, and not as an input buffer.
Fixes ticket #7188.
Signed-off-by: Marton Balint <cus@passwd.hu>
This does require libmysofa with today's latest commit (08f243d1ec).
They already had a pkg-config file, but the dependencies weren't setup right. Until now.
This should be included as `<lilv/lilv.h>`, same as is done in af_lv2.c.
Forcing the extra lilv-0 breaks platforms where the include dir is
`/usr/include/lilv/lilv.h` rather than
`/usr/include/lilv-0/lilv/lilv.h`.
The new include path works for both, because the `pkg-config --cflags`
includes `-I/usr/include/lilv-0`.
Fixes PMT parsing in some mpegts streams which contain
multiple tables within the PMT pid. Previously, the parser
assumed only one table was present in each packet, and discarded
the rest of the section data after attempting to parse the first
table.
A similar issue was documented in the BeyondTV software[1], which
helped me diagnose the same bug in the ffmpeg mpegts demuxer. I also
tried DVBInspector, libdvbpsi's dvbinfo, and tstools' tsinfo to
help debug. The former two properly read PMTs with multiple tables,
whereas the last has the same bug as ffmpeg.
I've created a minimal sample[2] which contains the combined PMT.
Here's what ffmpeg probe shows before and after this patch:
Before:
Input #0, mpegts, from 'combined-pmt-tids.ts':
Duration: 00:00:01.08, start: 4932.966167, bitrate: 741 kb/s
Program 1
No Program
Stream #0:0[0xf9d]: Audio: ac3, 48000 Hz, mono, fltp, 96 kb/s
Stream #0:1[0xf9b]: Audio: mp3, 0 channels, fltp
Stream #0:2[0xf9c]: Unknown: none
After:
Input #0, mpegts, from 'combined-pmt-tids.ts':
Duration: 00:00:01.11, start: 4932.966167, bitrate: 718 kb/s
Program 1
Stream #0:0[0xf9b]: Video: mpeg2video ([2][0][0][0] / 0x0002), none(tv, top first), 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc
Stream #0:1[0xf9c](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 384 kb/s
Stream #0:2[0xf9d](spa): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, mono, fltp, 96 kb/s
With the patch, the PMT is parsed correctly so the streams are
created in the correct order, are associated with "Program 1",
and their codecs are set correctly.
[1] http://forums.snapstream.com/vb/showpost.php?p=343816&postcount=201
[2] https://s3.amazonaws.com/tmm1/combined-pmt-tids.ts
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This allows remuxing streams from one mpegts container to another,
without requiring avformat_find_stream_info() (or using `ffmpeg
-probesize 32` on the cli).
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
SMPTE 386M (D-10) lists 4 as value to be used
SMPTE 377-1-2009 says
"The definitions of 00h (coSiting) and 04h (Rec 601) are equivalent. The value of 04h is deprecated. New
MXF encoders shall use the value of 00h instead."
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Adds tests for the hue angle and brightness filter parameters.
Renames the existing saturation parameter test for consistency.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
The artificial sample file sei-1.h264 contains five frames (IDR P B I B)
and the following SEI message types:
* Buffering period
* Picture timing
* Pan-scan rectangle (display as 4:3)
* User data registered, containing A/53 closed captions (captions match
frame content, including reordering)
* Recovery point (at the I frame)
* Display orientation (identity transformation)
* Mastering display (with arbitrary contents)
* Undefined SEI type 1234 (containing ascending bytes)
Formats ARGB32, XRGB32, ABGR32, and XBGR32 were added to V4L2 instead
of ill-defined deprecated RGB32/BGR32 pixel formats.
When pixel format is not specified explicitly FFmpeg tries formats in
order in which they are stored in the table. Therefore formats are
sorted as follows: BGR is preferred over RGB and XBGR is preferred
over ARGB, because it could give better performance by ignoring alpha
component.
'-sei xxx' is added to control SEI insertion, so far only mastering
display colour volume is available for testing.
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
Similar to H264, cbs_h265_{read, write}_nal_unit() can handle HEVC
prefix SEI NAL units. Currently mastering display colour volume SEI
message is added only, we may add more SEI message if needed later
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
Every fourcc in vaapi_drm_format_map should be in
vaapi_format_map, so add an assert to ensure we have the right maps.
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
The flag of input_available must be set when pic_start is not NULL, so
add an assert to ensure it is true. In addition, the assert on last_pic
is unnecessary now, so remove this assert.
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
The main issue here was the use of [i] instead of [i * 3] for the 32x32
matrix. As part of fixing this, I changed the code to match that used
in vdpau_hevc, which I spent a lot of time verifying.
I also changed to calculating NumPocTotalCurr using the existing helper,
which is what vdpau does.
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
fix the warning: "function declaration isn’t a prototype", in C
int foo() and int foo(void) are different functions. int foo()
accepts an arbitrary number of arguments, while int foo(void) accepts 0
arguments.
Signed-off-by: Jun Zhao <mypopydev@gmail.com>
Replaces the data pointers with the mapped cuvid ones.
Adds buffer_refs to the frame to ensure the needed contexts stay alive
and the cuvid idx stays allocated.
Adds another buffer_ref to unmap the frame when it's unreferenced itself.
This reverts commit 7d4e1f7cfb.
Accidentially pushed this with a batch of other patches, and it didn't
seem to break anything, so I went with it.
Except it does, so reverting it it is.
This mimics the logic flow in all the other callbacks
(pat_cb, sdt_cb, m4sl_cb), and avoids calling skip_identical()
for non PMT_TID packets.
Since skip_identical modifies internal state like
MpegTSSectionFilter.last_ver, this change prevents unnecessary
reprocessing on some streams which contain multiple tables in
the PMT pid. This can be observed with streams from certain US
cable providers, which include both tid=0x2 and another unspecified
tid=0xc0.
Signed-off-by: Aman Gupta <aman@tmm1.net>
No longer required since 63d875772d. The equivalent hack
for h264 was removed in that commit, but this one was missed.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Removes unnecessary data copies, and partially fixes potential issues
with dangling references held in said lists.
Reviewed-by: Mark Thompson <sw@jkqxz.net>
Signed-off-by: James Almer <jamrial@gmail.com>
This helps figuring out where the filter is slow:
70.53% ffmpeg_g ffmpeg_g [.] nlmeans_slice
25.73% ffmpeg_g ffmpeg_g [.] compute_safe_ssd_integral_image_c
1.74% ffmpeg_g ffmpeg_g [.] compute_unsafe_ssd_integral_image
0.82% ffmpeg_g ffmpeg_g [.] ff_mjpeg_decode_sos
0.51% ffmpeg_g [unknown] [k] 0xffffffff91800a80
0.24% ffmpeg_g ffmpeg_g [.] weight_averages
(Tested with a large image that takes several seconds to process)
Since this function is irrelevant speed wise, the file's TODO is
updated.
before: ssd_integral_image_c: 49204.6
after: ssd_integral_image_c: 44272.8
Unrolling by 4 made the biggest difference on odroid-c2 (aarch64);
unrolling by 2 or 8 both raised 46k cycles vs 44k for 4.
Additionally, this is a much better reference when writing SIMD (SIMD
vectorization will just target 16 instead of 4).
SIMD code will not have to deal with padding itself. Overwriting in that
function may have been possible but involve large overreading of the
sources. Instead, we simply make sure the width to process is always a
multiple of 16. Additionally, there must be some actual area to process
so the SIMD code can have its boundary checks after processing the first
pixels.
Most decoders (pgssubdec, ccaption_dec) are using -1 or UINT32_MAX for a
subtitle event which should be cleared at the next event.
Signed-off-by: Marton Balint <cus@passwd.hu>
This is large enough for all jpeg2000 files i tested. If some need more then this
should be changed to dynamic allocation. Dynamic allocation would need to be done
carefully as these are many relatively small arrays so repeatly reallocating them
would not be good.
The decrease is a clean and simple solution assuming it works for all files.
Fixes: OOM
Fixes: 6534/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_JPEG2000_fuzzer-4821490731057152
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It exists, so why not use it? Helps one get rid of additional
search path related flags in addition to PKG_CONFIG_{PATH,LIBDIR}
when utilizing a cross-prefix separate from the sysroot.
Older iOS devices don't have a hardware HEVC decoder, but the
software decoder offered by VideoToolbox is well-optimized and
performs much better than the ffmpeg decoder.
Signed-off-by: Aman Gupta <aman@tmm1.net>
The patch enables dynamic bitrate through ReconfigureEncoder method
from nvenc API.
This is useful for live streaming in case of network congestion.
Signed-off-by: pkviet <pkv.stream@gmail.com>
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
If there is input like DVB-T streams it can change aspect ratio
on-the-fly, so nvenc should respect this change and change aspect ratio
in encoder.
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
The thread id was invalid because it was not initialised
during the calls to init_complex_filtergraph.
This adds a flag to check for initialisation before trying to
peform the join.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Kevin Wheatley <kevin.j.wheatley@gmail.com>
I tested the previous mediacodec changes on seven different Android
TV devices, with both mpeg2 and h264 content. All except one worked
as expected. The exception was the MiBox3 running Android 6.0.1,
where playback would freeze on a frame every few seconds. I tested
two other AMLogic devices with newer Android versions that did not
show the same problem. H264 decoding on the MiBox3 was also not affected,
so this workaround applies only to OMX.amlogic.mpeg2.decoder.awesome
on Android API22.
There is a rumor that Xiaomi is planning to release Android Oreo for
the MiBox3, so I will revisit in a few months to confirm whether this
is specific to os/driver version or the chipset used in that device.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com>
The output_buffer_count==0 special case is no longer required, and
can cause spurious EAGAIN to surface to the user when input buffers
are filled up. Since the caller now knows if the decoder is accepting
new input (via current_input_buffer>=0), let the wait parameter
control whether we block or not.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com>
The new logic follows a recommendation by @rcombs to use
dequeueInputBuffer with a timeout of 0 as a way to detect
whether the codec wants more data. The dequeued buffer index is
kept in MediaCodecDecContext until it can be used next.
A similar technique is also used by the Google's official media
player Exoplayer: see MediaCodecRenderer.feedInputBuffer().
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com>
This can "demux" .vpy files. Autodetection of .vpy scripts is
intentionally not done, because it would be a major security issue. You
need to force the format, for example with "-f vapoursynth" for the
FFmpeg CLI tools.
Some minor code copied from other LGPL parts of FFmpeg.
I did not find a good way to test a few of the more obscure VS features,
like VFR nodes, compat pixel formats, or nodes with dynamic size/format
changes. These can be easily implemented on demand.
This code will print a warning if any user agent is set - even if the
API user used the proper non-deprecated "user_agent" option.
This change should not even break anything, because even if the user
sets the deprecated "user-agent" option, http.c copies it to the
"user_agent" option anyway.
If the API user doesn't set avg_frame_rate, matroskaenc will write the
current timebase as "default duration" for the video track. This makes
no sense, because the "default duration" implies the framerate of the
video. Since the timebase is forced to 1/1000, this will make the
resulting file claim 1000fps.
Drop it and don't write the element. It's optional, so it's better not
to write it if the framerate is unknown.
Strangely does not require FATE changes.
Enables one to test possibly nonstandard formats such as Opus or
FLAC in ISOBMFF, among other things.
This becomes much more useful if output segment format becomes an
option, or if the WebM segment feature gets removed.
It has not ever been working and has not been validated,
Additionally, mention that the segment file names should be changed
to end with webm instead of m4s, which is utilized for ISOBMFF
fragments.
Fixes stream field order written by avformat_write_header when "top"
option is specified on the command-line.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Uses the same mechanism as other codecs - conformance test files are
passed through the metadata filter (which, with no options, reads the
input and writes it back) and the output verified to match the input.
Create a buffer from the data instead and use the buffer destructor to free the
DeckLink frame. This avoids a memcpy of the frame data.
Signed-off-by: Marton Balint <cus@passwd.hu>
Temporarily keep the old method for ffmpeg_filters.c choose_pix_fmt and
avfiltergraph.c pick_format() until a paletted pixel format without alpha is
introduced.
Signed-off-by: Marton Balint <cus@passwd.hu>
The specs says that the the first color component in the color array is
not alpha, but simply 0.
Fixes 0 alpha of fate-suite/cvid/catfight-cvid-pal8-partial.mov
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes: runtime error: signed integer overflow: 2147483637 + 128 cannot be represented in type 'int'
Fixes: 6701/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WAVPACK_fuzzer-5358324934508544
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: runtime error: signed integer overflow: 2147483531 + 16384 cannot be represented in type 'int'
Fixes: 6615/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WAVPACK_fuzzer-5165715515506688
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In vc1_decode_i_blocks_adv mquant needs to be reset to its default value for
each macroblock, instead of once at the beginning of the slice.
DQPROFILE specifies which macroblocks can have an alternative quantizer step
size. When DQPROFILE specifies edges, the selection is applicable to the edges
of the picture. Slice edges are not selected by DQPROFILE.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Useful when transcoding videos at 29.97 fps because delivers a more accurate result for monitoring.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The -benchmark and -benchmark_all options now show user, system, and real time,
instead of just user time.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
1. an audio component with an ISO_639_language_descriptor in the PMT with the
audio_type field set to 0x03
2. a supplementary_audio_descriptor with the editorial_classification field set
to 0x01
3. an ac-3_descriptor or an enhanced_ac-3_descriptor with a component_type field
with the service_type flags set to Visually Impaired
Tested-by: Łukasz Krzciuk <lkrzciuk@vewd.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This way, every CodedBitstreamType->split_fragment() function can
safely assume the fragment passed to them will be reference counted,
potentially simplifying code.
Reviewed-by: Mark Thompson <sw@jkqxz.net>
Signed-off-by: James Almer <jamrial@gmail.com>
Overlap filtering of the first row and first column must be guarded
for out of bounds access of v->over_flags_plane.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
The track's media duration from the mdhd atom takes precedence
over both the stts and elst atom for calculating and setting
the track's total duraion.
Technically, we shouldn't be using the stts atom at all for
calculating stream durations.
This fixes incorrect stream and final packet durations on files
with edit lists that are longer than the media duration.
The FATE changes are expected, and output is more correct (the
AAC frame is not 1028 samples).
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
The av_rc4_crypt() documentation allows src == dst.
Silences the following warning:
libavformat/rtmpcrypt.c:304:36: warning: passing argument 2 of 'av_rc4_crypt' discards 'const' qualifier from pointer target type
Reported-by: Reino Wijnsma
There is a separate muxer(webmdashenc.c) for supporting VP9+webm output in DASH.
Hence in this muxer we will focus on supporting VP9 in MP4
Have verified playout support of VP9+MP4 in Chrome and Firefox.
Fixes: runtime error: signed integer overflow: 197710 * 10923 cannot be represented in type 'int'
Fixes: 7010/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-5667127596941312
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
HALFQP should only be added to the inverse quantizer when the block is
coded with PQUANT. When PQUANT is equal to ALTPQUANT, the original test
for the addition of HALFQP fails. A negative value for mquant indicates
that the value was derived from VOPDQUANT.
Fixes#4372
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
HALFQP should only be added to the inverse quantizer when the block is
coded with PQUANT. See 8.1.3.8 in VC-1 spec.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
mspel indicates the use of bicubic interpolation. The check wrongly included
MVMODE MV_PMODE_1MV_HPEL as using bilinear interpolation.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Add previously omitted overlap smooting and loop filtering for
frame/field-interlace pictures. For progressive pictures switch to the
re-implemented versions of overlap smooting and loop filtering.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
The existing implementation only used vc1_put_signed_blocks_clamped for I and
BI frames. This rewritten version is also applicable to P frame both
progressive and frame/field-interlace.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
The existing implementation did out-of-bounds reference pixel replication for
progressive reference frames. In interlaced reference frames both the even and
odd line on the horizontal edges need to be replicated.
Fixes#3262.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
According to VC-1 spec table 74, the last value in ff_vc1_dqscale should be
0x1041 instead of 0x1000.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
According to VC-1 spec 10.7.3.4, FIELDTX shall be set to the same type as the
motion vector for zero-coded blocks.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
The loop filter for P interlace field pictures needs the reference field type.
For luma, the reference field type was already available. Store the reference
field type for color-difference as well.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
The existing implementation did loop filtering for progressive
frames only. This rewritten version implements loop filtering for
all applicable frame types for both progessive and
frame/field-interlace.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
The new overlap smooting filter smoothes image pixels stored in v->block.
Switch to v->block instead of s->block for storing decoded image pixels for P
frames. Additionally, we must take incrementing *_blk_idx out of the
vc1_put_signed_blocks_clamped function.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
The existing implementation did overlap smoothing for progressive
frames only. This rewritten version implements overlap smoothing
for all applicable frame types for both progessive and
frame/field-interlace.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
before:
419022 decicycles in assemble_fragment, 2047 runs, 1 skips
after:
104621 decicycles in assemble_fragment, 2045 runs, 3 skips
Benched with a 2 minutes long 720x480 DVD mpeg2 sample.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Also fixes a bug where EOS buffer was sent with incorrect
pts when not using surface generation.
Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Signed-off-by: Aman Gupta <aman@tmm1.net>
As of 2a0eb8685, ff_mediacodec_dec_is_flushing() only returns
true in delay_flush mode. Make this more obvious by adding
delay_flush to the if statement.
Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Signed-off-by: Aman Gupta <aman@tmm1.net>
also fixes: runtime error: index 1456 out of bounds for type 'int16_t [16]'
Found-by: durandal_1707
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The crc flag is only stored since version 3 thus before this crcs do not
work. We increase the version as needed same as we do with pix_fmts
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Interlaced frame pictures do not contain the MVMODE or MVMODE2 bitstream
element. Trying to parse this element and passing a nonzero value to the
hardware decoder results in small inaccuracies in the decoded picture.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Should be useful for muxers that require values as defined in the
vpcc atom but don't need to write the atom itself.
Signed-off-by: James Almer <jamrial@gmail.com>
When use new decode APIs(avcodec_send_packet/avcodec_receive_frame),
don't need to setting the deprecated field refcounted_frames.
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Jun Zhao <mypopydev@gmail.com>
When use new decode APIs(avcodec_send_packet/avcodec_receive_frame),
don't need to setting the deprecated field refcounted_frames.
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Jun Zhao <mypopydev@gmail.com>
When use new decode APIs(avcodec_send_packet/avcodec_receive_frame),
don't need to setting the deprecated field refcounted_frames.
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Jun Zhao <mypopydev@gmail.com>
scaleforsame_y references ref_field_type. Therefore, it needs to be set
before scaleforsame is called.
Fixes#2557.
Signed-off-by: Jerome Borsboom <jerome.borsboom@carpalis.nl>
Fixes a warning:
libavformat/dashdec.c:1900:65: warning: argument to 'sizeof' in 'memcpy' call is the same pointer type 'struct fragment *' as the destination; expected 'struct fragment' or an explicit length
For filters based on framesync, the input frame was managed
by framesync, so we should not directly keep and destroy it,
instead we make a clone of it here, or else double-free will occur.
But for other filters not based on framesync, we still need to
free the input frame inside filter_frame.
Signed-off-by: Ruiling Song <ruiling.song@intel.com>
The existing version which was cherry-picked from Libav does not work
with FFmpeg framework, because ff_request_frame() was totally
different between Libav (recursive) and FFmpeg (non-recursive).
The existing overlay_qsv implementation depends on the recursive version
of ff_request_frame to trigger immediate call to request_frame() on input pad.
But this has been removed in FFmpeg since "lavfi: make request_frame() non-recursive."
Now that we have handy framesync support in FFmpeg, so I make it work
based on framesync. Some other fixing which is also needed to make
overlay_qsv work are put in a separate patch.
Signed-off-by: Ruiling Song <ruiling.song@intel.com>
This doesn't support saio atoms with more than one offset.
Signed-off-by: Jacob Trimble <modmaker@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Similar to 4c9c4fe8b2, but for durations. This fixes#7151, where
the report duration and bitrate on a mpegts stream is wildly off
due to the dvb_teletext stream's timings.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In write only mode, the TCP receive buffer's data keeps growing with
http response messages and the buffer eventually becomes full.
This results in zero tcp window size, which in turn causes unwanted
issues, like, terminated tcp connection. The issue is apparent when
http persistent connection is enabled in hls/dash live streaming use
cases. To overcome this issue, the logic here reads the buffer data
when a file transfer is completed, so that any accumulated data in
the recieve buffer gets flushed out.
reference hls support fmp4 file from draft-pantos-http-live-streaming-20
the spec describes version 7 of hls protocol
Suggested-by: Ronak <ronak2121@yahoo.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
The headers from where the dimensions are read in actual files
are limited to 16bit per component.
Fixes: Timeout
Fixes: 6305/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DFA_fuzzer-4824270749302784
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This refactors get_cookies to simplify some code paths, specifically for
skipping logic in the while loop or exiting it. It also simplifies the logic
for appending additional values to *cookies by replacing strlen/malloc/snprintf
with one call av_asnprintf.
This refactor fixes a bug where the cookie_params AVDictionary would get leaked
if we failed to allocate a new buffer for writing to *cookies.
Branch to global symbol results in reference to PLT, and when compiling
for THUMB-2 - in a R_ARM_THM_JUMP19 relocation. Some linkers don't
support this relocation (ld.gold), while others can end up truncating
the relocation to fit (ld.bfd).
Convert this branch through PLT into a direct branch that the assembler
can resolve locally.
See https://github.com/android-ndk/ndk/issues/337 for background.
The current workaround is to disable neon during gstreamer build,
which is not optimal and can be reverted after this patch:
41556c4157
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes frame corruption issue when decoder started reusing frames
while they are still in use of encoding process
Issue with frame corruption was reproduced using:
avconv.exe -y -hwaccel d3d11va -hwaccel_output_format d3d11 -i input.h264 -an -c:v h264_amf output.mkv
It is recommended to use -extra_hw_frames 16 option in case if hw frames
number in pool is not enough
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
- enable the parsing code
- use the new buffer instead of replacing the context one
- do not push/pop configuration, just discard the exiting one
- propagate errors correctly
Without properly grouping the checks, the second test would execute for
MSVC cl.exe, which results in configure getting stuck since cl.exe -? is
an interactive paginated help screen, waiting for input.
Remove the wincrypt API calls since we don't support XP anymore and
bcrypt is available since Vista, even on Windows Store builds.
Signed-off-by: Martin Storsjö <martin@martin.st>
- Parse schm atom to get different encryption schemes.
- Allow senc atom to appear in track fragments.
- Allow 16-byte IVs.
- Allow constant IVs (specified in tenc).
- Allow only tenc to specify encryption (i.e. no senc/saiz/saio).
- Use sample descriptor to detect clear fragments.
This doesn't support:
- Different sample descriptor holding different encryption info.
- Only first sample descriptor can be encrypted.
- Encrypted sample groups (i.e. seig).
- Non-'cenc' encryption scheme when using -decryption_key.
Signed-off-by: Jacob Trimble <modmaker@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Show a more useful error message which specifies the required driver version
for the build, and use the correct context in the error message for WIN32.
Signed-off-by: Marton Balint <cus@passwd.hu>
Silences several warnings:
libavcodec/dxva2_internal.h:107:98: warning: pointer type mismatch in conditional expression
libavcodec/dxva2_internal.h:109:94: warning: pointer type mismatch in conditional expression
Reported-by: Reino Wijnsma
Fixes the following warnings:
libavdevice/vfwcap.c:331:35: warning: passing argument 1 of 'av_parse_video_size' from incompatible pointer type
libavdevice/vfwcap.c:331:59: warning: passing argument 2 of 'av_parse_video_size' from incompatible pointer type
Reported-by: Reino Wijnsma
SDL_QueryTexture and SDL_DestroyTexture require that the input texture
pointer be non-null. Debug builds of SDL will correctly check for this
and break program execution. This patch fixes this by checking the
status of the texture pointer.
Signed-off-by: Matt Oliver <protogonoi@gmail.com>
Fixes https://trac.ffmpeg.org/ticket/2798
This makes movenc handle AV_DISPOSITION_ATTACHED_PIC and write
the associated pictures in iTunes cover atom. This corresponds
to how 'mov' demuxer parses and exposes the cover images when
reading.
Most of the existing track handling loops properly ignore
these 'virtual streams' as MOVTrack->entry is never incremented
for them. However, additional tests are added as needed to ignore
them.
Tested to produce valid output with:
ffmpeg -i movie.mp4 -i thumb.jpg -disposition:v:1 attached_pic \
-map 0 -map 1 -c copy movie-with-cover.mp4
The cover image is also copied correctly with:
ffmpeg -i movie-with-cover.mp4 -map 0 -c copy out.mp4
AtomicParseley says that the attached_pic stream is properly
not visible in the main tracks of the file.
Signed-off-by: Timo Teräs <timo.teras@iki.fi>
av_guess_sample_aspect_ratio() will return undefined or missing
value as {0,1}. This fixes show_stream() to check numerator to
display 'N/A' when appropriate. show_frame() does this already
correctly.
Signed-off-by: Timo Teräs <timo.teras@iki.fi>
The logic is applicable only when use_template is enabled and use_timeline
is disabled. The logic monitors the flow of segment indexes. If a streams's
segment index value is not at the expected real time position, then
the logic corrects that index value.
Typically this logic is needed in live streaming use cases. The network
bandwidth fluctuations are common during long run streaming. Each
fluctuation can cause the segment indexes fall behind the expected real
time position. Without this logic, players will not be able to consume
the content, even after encoder's network condition comes back to
normal state.
availability time of Nth segment = availabilityStartTime + (N*segment duration) - availabilityTimeOffset.
This field helps to reduce the latency by about a segment duration in streaming mode.
@availabilityStartTime specifies the anchor for the computation of the earliest
availability time (in UTC) for any Segment in the Media Presentation.
As per this requirement, the @AvailabilityStartTime should be set to the
wallclock time at which the first frame of the first segment begins encoding.
But, it was getting set only when the first segment was completely ready. Making
the required correction in this patch. This correction is mainly needed to reduce
the latency in live streaming use cases.
Calling 'write_manifest' from 'write_header' was causing creation of
first MPD with invalid values. Ex: zero @duration param value. Also,
the manifest files (MPD or M3U8s) should be created when at-least
one media frame is ready for consumption.
When use_template is enabled and use_timeline is disabled, typically
it is required to generate the segments at the configured segment duration
rate on an average. This commit is particularly needed to handle the
segmentation when video frame rates are fractional like 29.97 or 59.94 fps.
There are use cases where average segment duration needs to be configured
and muxer is expected to maintain the average segment duration. So, using
the name 'min_seg_duration' will be misleading. So, changing the parameter
name to 'seg_duration', where it can be minimum segment duration or average
segment duration based on the use-case. The additional updates needed for
this functinality are made the sub-sequent patches of this patch series.
The HLSContext struct contains fields which duplicate the data stored in the
avio_opts field. This change removes those fields in favor of avio_opts, and
updates the code accordingly.
The original patch caused the buffer pointed to by new_cookies in open_url to be
leaked. The only thing that buffer is used for is to store the value until it
can be passed to av_dict_set. To fix the leak, v2 of the patch simply calls
av_dict_set with the AV_DICT_DONT_STRDUP_VAL flag, so that the dictionary takes
ownership of the memory instead of copying it again.
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Richard Shaffer <rshaffer@tunein.com>
The rw_timeout option is currently not applied when opening media playlist,
segment, or encryption key URLs. This can cause the HLS demuxer to block
indefinitely, even when the rw_timeout option has been specified. This change
simply enables carrying over the rw_timeout option when the demuxer opens these
URLs.
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Richard Shaffer <rshaffer@tunein.com>
The rw_timeout option is currently not applied when opening media playlist,
segment, or encryption key URLs. This can cause the HLS demuxer to block
indefinitely, even when the rw_timeout option has been specified. This change
simply enables carrying over the rw_timeout option when the demuxer opens these
URLs.
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Richard Shaffer <rshaffer@tunein.com>
This prevents creating potentially broken files, as both the AV1 and
the AV1 in ISOMBFF specs are unfinished.
Signed-off-by: James Almer <jamrial@gmail.com>
Some old mingw-w64 builds seem to provide an incomplete implementation
of the API. Add an extra check to make sure it's disabled for those.
Signed-off-by: James Almer <jamrial@gmail.com>
@@ -152,8 +152,11 @@ static int subband_bufer_alloc(DCAEncContext *c)
staticvoidsubband_bufer_free(DCAEncContext*c)
{
int32_t*bufer=c->subband[0][0]-DCA_ADPCM_COEFFS;
av_freep(&bufer);
if(c->subband[0][0]){
int32_t*bufer=c->subband[0][0]-DCA_ADPCM_COEFFS;
av_free(bufer);
c->subband[0][0]=NULL;
}
}
staticintencode_init(AVCodecContext*avctx)
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